• Title/Summary/Keyword: Error Reduction

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Thermal Aware Buffer Insertion in the Early Stage of Physical Designs

  • Kim, Jaehwan;Ahn, Byung-Gyu;Kim, Minbeom;Chong, Jongwha
    • JSTS:Journal of Semiconductor Technology and Science
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    • v.12 no.4
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    • pp.397-404
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    • 2012
  • Thermal generation by power dissipation of the highly integrated System on Chip (SoC) device is irregularly distributed on the intra chip. It leads to thermal increment of the each thermally different region and effects on the propagation timing; consequently, the timing violation occurs due to the misestimated number of buffers. In this paper, the timing budgeting methodology considering thermal variation which contains buffer insertion with wire segmentation is proposed. Thermal aware LUT modeling for cell intrinsic delay is also proposed. Simulation results show the reduction of the worst delay after implementing thermal aware buffer insertion using by proposed wire segmentation up to 33% in contrast to the original buffer insertion. The error rates are measured by SPICE simulation results.

Vowel Recognition Using the Fractal Dimensioin (프랙탈 차원을 이용한 모음인식)

  • 최철영
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1994.06c
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    • pp.364-367
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    • 1994
  • In this paper, we carried out some experiments on the Korean vowel recognition using the fractal dimension of the speech signals. We chose the Mincowski-Bouligand dimensioni as the fractal dimension, and computed it using the morphological covering method. For our experiments, we used both the fractal dimension and the LPC cepstrum which is conventionally known to be one of the best parameters for speech recognition, and examined the usefulness of the fractal dimension. From the vowel recognition experiments under various consonant contexts, we achieved the vowel recognition error rats of 5.6% and 3.2% for the case with only LPC cepstrum and that with both LPC cepstrum and the fractal dimension, respectively. The results indicate that the incorporation of the fractal dimension with LPC cepstrum gies more than 40% reduction in recognition errors, and indicates that the fractal dimension is a useful feature parameter for speech recognition.

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Weighted Distance-Based Quantization for Distributed Estimation

  • Kim, Yoon Hak
    • Journal of information and communication convergence engineering
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    • v.12 no.4
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    • pp.215-220
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    • 2014
  • We consider quantization optimized for distributed estimation, where a set of sensors at different sites collect measurements on the parameter of interest, quantize them, and transmit the quantized data to a fusion node, which then estimates the parameter. Here, we propose an iterative quantizer design algorithm with a weighted distance rule that allows us to reduce a system-wide metric such as the estimation error by constructing quantization partitions with their optimal weights. We show that the search for the weights, the most expensive computational step in the algorithm, can be conducted in a sequential manner without deviating from convergence, leading to a significant reduction in design complexity. Our experments demonstrate that the proposed algorithm achieves improved performance over traditional quantizer designs. The benefit of the proposed technique is further illustrated by the experiments providing similar estimation performance with much lower complexity as compared to the recently published novel algorithms.

Improvement of a Pound-Drever-Hall Technique to Measure Precisely the Free Spectral Range of a Fabry-Perot Etalon

  • Seo, Dong-Sun;Park, Chongdae;Leaird, Daniel E.;Weiner, Andrew M.
    • Journal of the Optical Society of Korea
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    • v.19 no.4
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    • pp.357-362
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    • 2015
  • We examine the principle of a modified Pound-Drever-Hall (PDH) technique to measure the free spectral range of a Fabry-Perot etalon (FPE). The FPE's periodic transmission of phase-modulated light allows us to adopt a sampling theorem to develop a new relationship for the PDH error signal. This leads us to find the key parameters governing the measurement accuracy: the phase modulation index ${\beta}$ and the FPE finesse. Without any additional complexity for background noise reduction, we achieve a measurement accuracy of 0.5 ppm. The improvement is mainly attributed to the wide-band phase modulation approaching ${\beta}=10$, and partly to the use of both reflected and transmitted light from the FPE and good FPE finesse.

Application of Surrogate Modeling to Design of A Compressor Blade to Optimize Stacking and Thickness

  • Samad, Abdus;Kim, Kwang-Yong
    • International Journal of Fluid Machinery and Systems
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    • v.2 no.1
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    • pp.1-12
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    • 2009
  • Surrogate modeling is applied to a compressor blade shape optimization to modify its stacking line and thickness to enhance adiabatic efficiency and total pressure ratio. Six design variables are defined by parametric curves and three objectives; efficiency, total pressure and a combined objective of efficiency and total pressure are considered to enhance the performance of compressor blade. Latin hypercube sampling of design of experiments is used to generate 55 designs within design space constituted by the lower and upper limits of variables. Optimum designs are found by formulating a PRESS (predicted error sum of squares) based averaging (PBA) surrogate model with the help of a gradient based optimization algorithm. The optimum designs using the current variables show that, to optimize the performance of turbomachinery blade, the adiabatic efficiency objective is improved substantially while total pressure ratio objective is increased a very small amount. The multi-objective optimization shows that the efficiency can be increased with the less compensation of total pressure reduction or both objectives can be increased simultaneously.

A Nonlinear Filtered-X LMS Algorithm for the Nonlinear Compensation of the Secondary Path in Active Noise Control (능동 소음 제어 시스템의 2차 경로 비선형 특성을 보상하기 위한 적응 비선형 Filtered-X Least Mean Square (FX-LMS) 알고리듬)

  • Jeong, I.S.;Kim, D.H.;Nam, S.W.
    • Proceedings of the KIEE Conference
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    • 2004.11c
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    • pp.565-567
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    • 2004
  • In active noise control (ANC) systems, the convergence behavior of the conventional Filtered-X Least Mean Square (FXLMS) algorithm may be affected by nonlinear distortions in the secondary path (e.g., in the power amplifiers, loudspeakers, transducers, etc.), which may lead to degradation of the error-reduction performance of the ANC systems. In this paper, a stable FXLMS algorithm with fast convergence is proposed to compensate for undesirable nonlinear distortions in the secondary-path of ANC systems by employing the Volterra filtering approach. In particular, the proposed approach is based on the utilization of the conventional P-th order inverse approach to nonlinearity compensation in the secondary path of ANC systems. Finally, the simulation results showed that the proposed approach yields a better convergence behavior In the nonlinear ANC systems than the conventional FXLMS.

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Actual Research on the Estimation Technique of the Future Trip in Pusan City. (부산시장래교통량의 추계수법에 관한 실증적 연구)

  • 오윤표
    • Journal of Korean Society of Transportation
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    • v.5 no.2
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    • pp.97-112
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    • 1987
  • The objective of this study is to construct not only trip production and attraction in Pusan but also to study and examine appropriateness of the model positively. Depending on the estimation models of trip production and attraction of each zone that have been constructed in this study, it has been proved that the formula of multiple regression by the explanation variables like the indices of total employees, total students, floor spaces of residentials and floor spaces of educational and cultural areas within the study areas have very high explanatory capacity and appropriateness. It si considered that a study of method on new division, integration or omission etc. of the existing zones preceeding for reduction of calculation quantity and a study of estimation error have to be done for future study, if these models are used actually.

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Implementation of Automatic Microphone Volume Controller and Recognition Rate Improvement (자동 입력레벨 조절기의 구현 및 인식 성능 향상)

  • 김상진;한민수
    • Proceedings of the IEEK Conference
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    • 2001.09a
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    • pp.503-506
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    • 2001
  • In this paper, we describe the implementation of a microphone input level control algorithm and the speech improvement with this level controller in personal computer environment. The volume of speech obtained through a microphone affects the speech recognition rate directly. Therefore, proper input volume level control is desired fur better recognition. We considered some conditions for the successful volume controller implementation firstly, then checked its usefulness on our speech recognition system with common office environment speech database. Cepstral mean subtraction is also utilized far the channel-effect compensation of the database. Our implemented controller achieved approximately 50% reduction, i.e., improvement in speech recognition error rate.

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DLL Design of SMD Structure with DCC using Reduced Delay Lines (지연단을 줄인 SMD 구조의 DCC를 가지는 DLL 설계)

  • Hong, Seok-Yong;Cho, Seong-Ik;Shin, Hong-Gyu
    • The Transactions of The Korean Institute of Electrical Engineers
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    • v.56 no.6
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    • pp.1133-1138
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    • 2007
  • DLLs(Delay Locked Loops) have widely been used in many systems in order to achieve the clock synchronization. A SMD (Synchronous Mirror Delay) structure is used both for skew reduction and for DCC (Duty Cycle Correction). In this paper, a SMD based DLL with DCC using Reduced Delay Lines is proposed in order to reduce the clock skew and correct the duty cycle. The merged structure allows the forward delay array to be shared between the DLL and the DCC, and yields a 25% saving in the number of the required delay cells. The designed chip was fabricated using a $0.25{\mu}m$ 1-poly, 4-metal CMOS process. Measurement results showed the 3% duty cycle error when the input signal ranges from 80% to 20% and the clock frequency ranges from 400MHz to 600MHz. The locking operation needs 3 clock and duty correction requires only 5 clock cycles as feature with SMD structure.

Fast Speaker Adaptation Using Sub-Stream Based Eigenvoice (Sub-Stream 기반의 Eigenvoice를 이용한 고속 화자적응)

  • Song, Hwa-Jeon;Lee, Jong-Seok;Kim, Hyung-Soon
    • MALSORI
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    • v.55
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    • pp.93-102
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    • 2005
  • In this paper, sub-stream based eigenvoice method is proposed to overcome the weak points of conventional eigenvoice and dimensional eigenvoice. In the proposed method, sub-streams are automatically constructed by the statistical clustering analysis that uses the correlation information between dimensions. To obtain the reliable distance matrix from covariance matrix for dividing into optimal sub-streams, MAP adaptation technique is employed to the covariance matrix of training data and the sample covariance of adaptation data. According to our experiments, the proposed method shows $41\%$ error rate reduction when the number of adaptation data is 50.

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