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IEEE SOI-3D-Subthreshold Microelectronics Technology Unified Conference (S3S)

作者： S. Krishnan J.G. Fossum P.C. Yeh O. Faynot S. Cristoloveanu J. Gautier University of Florida Gainesville FL USA LETI DMEL-CENG France LPCS-ENSERG France IBM NY USA

Fully depleted (FD) SOI CMOS is a contender for low-voltage IC applications. However, as FD/SOI MOSFETs are scaled, floating-body effects, which previously seemed insignificant, become important. In this paper, we report kinks in the measured subthreshold current-voltage characteristics of highly scaled FD/SOI MOSFETs, and we describe and model the underlying physical mechanism, showing how it differs from the familiar kink effect in partially depleted (PD) devices. The insight afforded qualifies the meaning of FD/SOI and implies new design issues for low-voltage SOI CMOS.

关键词： MOSFETs Impact ionization Body regions Back linear predictive coding Application specific integrated circuits CMOS integrated circuits Current measurement Current-voltage characteristics Semiconductor device modeling

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A robust parametric method for signal reconstruction

A robust parametric method for signal reconstruction

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Asilomar Conference on Signals, Systems & Computers

作者： S.R. Pillai A. Antoniou Department of Electrical Engineering University of Victoria Victoria BC Canada

In the context of signal reconstruction and coding, a new robust parametric formulation to linear predictive coding (LPC) is introduced. The linear prediction filter coefficients are transformed into a set of weighted line frequencies. The positive weights play the dual role of a new set of parameters and simultaneously they exhibit the relative importance of the associated line frequencies. This new representation for LPC is shown to be always stable under quantization.

关键词： Robustness Signal reconstruction linear predictive coding Speech coding Speech analysis Distortion measurement Mean square error methods Polynomials Frequency Redundancy

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A comparison of SPOX and C implementations of a LPC vocoder on the TMS320C30 EVM

A comparison of SPOX and C implementations of a LPC vocoder ...

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IEEE Southeastcon

作者： D.A. Kilpatrick T.C. Jannett Department of Electrical and Computer Engineering University of Alabama Birmingham Birmingham AL USA

The paper describes a comparison of a C implementation of a linear predictive voice coder (LPC) and an implementation based on Spectron Microsystem's Signal Processing Operating System (SPOX). The hardware platform was a Texas Instruments TMS320C30 Evaluation Module. The SPOX and C implementations were compared based on execution time, ease of program development and maintenance, and portability to different hardware platforms. The vocoder algorithms and the results of the comparison of both implementations are presented.

关键词： linear predictive coding Vocoders Speech synthesis Speech analysis Filters Hardware Signal synthesis Synthesizers Humans Signal processing algorithms

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Correction to "Shower Counters for SLAG Experiments E142/E143" [Erratum]

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IEEE Transactions on Nuclear Science 1995年第6期42卷 2347-2347页

作者： H. Borel P. Grenier J. LeMeur R. Lombard J.C. Logol J. Marroncle J. Morgenstern J.P. Mouly B. Paul F. Staley Y. Terrien V. Breton C. Comptour M. Crouau H. Fonvielle Y. Roblin G. Savinel R. Gearhart

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Call for papers

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IEEE Transactions on Rehabilitation Engineering 1995年第4期3卷 373-373页

作者： K. Kawamura R.A. Peters S. Bagchi M. Iskarous M. Bishay Center for Intelligent Systems Vanderbilt University Nashville TN USA

Prospective authors are requested to submit new, unpublished manuscripts for inclusion in the upcoming event described in this call for papers.

关键词： Intelligent robots Intelligent systems Electrical engineering linear predictive coding Biographies Image processing Target recognition Automatic control Control systems

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predictive coding of Speech Using Microphone/Speaker Adaptation and Vector Quantization

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1994年第2期2卷 266-273页

作者： Aarskog, Alv I. Guren, Hans C. ABB Corp Res N-1361 Billingstad Norway

A problem with speech coders based on trained quantizers is their lack of robustness against variations in the microphone and input filter response. In this paper a simple backward adaptive prefiltering technique is proposed as a means of improving the robustness and quality of a speech coder at no cost in bit rate. The technique is particularly useful in conjunction with vector quantization (VQ) of the linear predictive coding (LPC) parameters. The performance of the prefilter, denoted a microphone and speaker adaptation (MSA) filter, has been evaluated in terms of prediction gain and spectral distortion, together with objective and subjective quality of a 7.5 kbit/s CELP speech coder. In this coder a 10-bit direct VQ of the LPC parameters using the residual energy distortion measure has been applied. This is consistent with the covariance method of LPC analysis. Simulation results illustrate that the MSA filter significantly improves the performance and robustness of the LPC VQ against changes in the input response. The 7.5 kbit/s CELP with a trained excitation codebook and MSA was found to be clearly better (subjectively and objectively) than the one without MSA. The coder with MSA also showed to be practically indistinguishable from the same CELP with unquantized LPC coefficients and a stochastic excitation codebook.

关键词： predictive coding Speech coding Microphones linear predictive coding Robustness Filters Costs Bit rate Vector quantization Performance gain

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TRELLIS EXCITATION SPEECH coding AT LOW BIT RATES

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IEEE TRANSACTIONS ON COMMUNICATIONS 1994年第2-4期42卷 1902-1910页

作者： KANG, SW FISCHER, TR WASHINGTON STATE UNIV SCH ELECT ENGN & COMP SCI DEPT ELECT & COMP ENGN PULLMAN WA 99164 USA

Trellis coded vector quantization (TCVQ) and code-excited linear prediction (CELP) coding are combined to form an efficient low-bit rate speech coding system. The resulting system uses a trellis search to select the synthesis filter excitation sequence, and is referred to as trellis excitation coding (TEC). Simulations are performed for encoding rates of 6.4 and 8 kbps. Informal listening tests indicate that the 8 kbps TEC system has quality roughly between that of 6-bit and 7-bit mu-law PCM with mu = 255. The 6.4 kbps TEC system provides speech quality between 5-bit and 6-bit mu-law PCM. A subjective comparison with vector sum excited linear prediction (VSELP) indicates that the 8 kbps TEC and the VSELP reconstructed speech is about equally preferable.

关键词： Speech coding Bit rate Encoding Vector quantization Phase change materials linear predictive coding Delay Speech synthesis Filters System testing

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A NEW ARTIFICIAL SPEECH SIGNAL FOR OBJECTIVE QUALITY EVALUATION OF SPEECH coding SYSTEMS

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IEEE TRANSACTIONS ON COMMUNICATIONS 1994年第2-4期42卷 664-672页

作者： ITOH, K KITAWAKI, N IRII, H NAGABUCHI, H NIPPON TELEGRAPH & TEL PUBL CORP TELECOMMUN NETWORKS LABS TOKYO 190 JAPAN

This paper describes a new artificial speech signal (ASVQ: Artificial Speech by Vector Quantization technique) which reflects the average characteristics of the human voice. The ASVQ is intended for use as a test signal in the objective evaluation of speech coding system quality. To obtain the average characteristics, a very large speech data base is analyzed. The ASVQ generation method which reflects the extracted average characteristics of the human voice is formulated. This method applies vector quantizing analysis to the speech data base. The LPC speech synthesis circuit is used to reproduce the average characteristics. Finally, the new artificial speech signal is compared with a human voice and the estimation accuracy of the subjective quality of speech coding systems and nonlinear distortions is evaluated.

关键词： Speech analysis Human voice Speech coding Vector quantization System testing Data analysis Character generation Data mining linear predictive coding Speech synthesis

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AN LSF QUANTIZER FOR THE NORTH-AMERICAN HALF-RATE SPEECH CODER

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IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY 1994年第3期43卷 413-419页

作者： DEMARCA, JRB AT&T BELL LABS SPEECH PROC RES DEPTMURRAY HILLNJ 07974

The design of the half-rate coder for the North-American cellular communication standard poses a challenging problem. The desired speech quality is that of the full-rate coder, but the total bit rate specified for the half-rate is only 6.5 kb/s. Since the mobile communication channel is very noisy, the use of error-correcting codes is necessary, which leaves only about 4 kb/s for the actual speech information. This very restricted bit budget requires that fewer than 30 b be allotted to the quantization of the LSF parameters, which precludes the use of scalar quantization. In this work, a quantizer design procedure which is tailored for the transmission frame structure of the North-American (TIA) cellular communication standard is presented. It is basically a split vector quantizer which makes use of interframe prediction to lower the number of bits required for quantization. Since the prediction is performed only on every other frame, the error propagation is limited. The quantization technique yields good performance in the 26-27 b/frame range, and the complexity of its implementation is less than 0.5 Mip.

关键词： Quantization Frequency Communication standards linear predictive coding Bit rate Filters Mobile communication Error correction codes Speech coding predictive models

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A Frobenius Norm Approach to Glottal Closure Detection from the Speech Signal

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1994年第2期2卷 258-265页

作者： Ma, Changxue Kamp, Yves Willems, Lei F. Eindhoven Univ Technol Inst Percept Res NL-5600 MB Eindhoven Netherlands

The detection of glottal closure instants has been a necessary step in several applications of speech processing, such as voice source analysis, speech prosody manipulation and speech synthesis. This paper presents a new algorithm for glottal closure detection that compares favorably with other methods available today in terms of robustness and computational efficiency. In this paper, we propose to use the singular value decomposition (SVD) approach to detect the instants of glottal closure from the speech signal. The proposed SVD method amounts to calculating the Frobenius norms of signal matrices and therefore is computationally efficient. Moreover, it produces well-defined and reliable peaks that indicate the instants of glottal closure. Finally, with the introduction of the total linear least squares technique, two proposed methods are reinvestigated and unified into the SVD framework.

关键词： Speech analysis Speech synthesis Speech processing Acoustic noise Singular value decomposition Filters Pulse modulation Speech coding linear predictive coding predictive models

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