• Title/Summary/Keyword: FIR 근사

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An Approximated RLS Algorithm for Adaptive Parameter Estimation (적응 파라미터 예측을 위한 근사화된 RLS 알고리즘)

  • Ahn, Bong-Man;Hwang, Jee-Won;Ryoo, Jung-Rae;Cho, Ju-Phil
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.9C
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    • pp.922-928
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    • 2007
  • This paper presents the fast adaptive algorithm which applies an approximation scheme into RLS algorithm. The proposed algorithm(D-RLS) derives a QRD RLS algorithm derivation process from RLS algorithm recursively. D-RLS has the similar pattern as the algorithm having the approximation that input signals are separated respectively. Computational complexity of D-RLS is O(N), fewer than $O(N^2)$. To evaluate performance of proposed algorithm, we use the system identification method of FIR and Volterra system. And, finally, we can show D-RLS has an excellent performance.

A Study on TSIUVC Approximate-Synthesis Method using Least Mean Square and Frequency Division (주파수 분할 및 최소 자승법을 이용한 TSIUVC 근사합성법에 관한 연구)

  • 이시우
    • Journal of Korea Multimedia Society
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    • v.6 no.3
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    • pp.462-468
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    • 2003
  • In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and an unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including Unvoiced Consonant) searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This paper present a new method of TSIUVC approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547KHz below and 2.813KHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TSIUVC. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.

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Learning-based Super-resolution for Text Images (글자 영상을 위한 학습기반 초고해상도 기법)

  • Heo, Bo-Young;Song, Byung Cheol
    • Journal of the Institute of Electronics and Information Engineers
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    • v.52 no.4
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    • pp.175-183
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    • 2015
  • The proposed algorithm consists of two stages: the learning and synthesis stages. At the learning stage, we first collect various high-resolution (HR)-low-resolution (LR) text image pairs, and quantize the LR images, and extract HR-LR block pairs. Based on quantized LR blocks, the LR-HR block pairs are clustered into a pre-determined number of classes. For each class, an optimal 2D-FIR filter is computed, and it is stored into a dictionary with the corresponding LR block for indexing. At the synthesis stage, each quantized LR block in an input LR image is compared with every LR block in the dictionary, and the FIR filter of the best-matched LR block is selected. Finally, a HR block is synthesized with the chosen filter, and a final HR image is produced. Also, in order to cope with noisy environment, we generate multiple dictionaries according to noise level at the learning stage. So, the dictionary corresponding to the noise level of the input image is chosen, and a final HR image is produced using the selected dictionary. Experimental results show that the proposed algorithm outperforms the previous works for noisy images as well as noise-free images.

A Study on a Searching, Extraction and Approximation-Synthesis of Transition Segment in Continuous Speech (연속음성에서 천이구간의 탐색, 추출, 근사합성에 관한 연구)

  • Lee, Si-U
    • The Transactions of the Korea Information Processing Society
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    • v.7 no.4
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    • pp.1299-1304
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    • 2000
  • In a speed coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and an unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including UnVoiced Consonant) searching, extraction ad approximation-synthesis method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This method based on a zerocrossing rate and pitch detector using FIR-STREAK Digital Filter. As a result, the extraction rates of TSIUVC are 84.8% (plosive), 94.9%(fricative), 92.3%(affricative) in female voice, and 88%(plosive), 94.9%(fricative), 92.3%(affricative) in male voice respectively, Also, I obain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547kHz below and 2.813kHz above. This method has the capability of being applied to speech coding of low bit rate, speech analysis and speech synthesis.

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Approximated Constrained Least Squares Filter for Real-Time Directionally Adaptive Image Restoration (제약적 최소 제곱 필터의 근사화를 이용한 실시간 방향 적응적 영상복원)

  • Cho, Changhun;Jeon, Jaehwan;Paik, Joonki
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.12
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    • pp.150-158
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    • 2013
  • In this paper we present approximated constrained least squares filter for real-time directionally adaptive image restoration. The proposed method makes a hardware implementation easier for real-time image restoration because of reducing the filter size. Furthermore, for directional adaptive image restoration, this paper estimates the local orientation by analyzing the covariance matrix and applies to approximated constrained least squares filter. Experimental results show that the proposed method is sharper and less artifacts than existing methods.

A Method of Designing Low-power Feedback Active Noise Control Filter for Headphones/Earphones (헤드폰/이어폰을 위한 저전력 피드백 능동 소음 제어 필터 설계 방법)

  • Seo, Ji-ho;Youn, Dae-Hee;Park, Young-Cheol
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.10 no.1
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    • pp.57-65
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    • 2017
  • This paper presented a method of designing low-power feedback active noise control filter optimized for headphones/earphones. Using constrained optimization, we obtained a high order FIR noise control filter to ensure reasonable noise attenuation performance at high sampling frequency environment. Then using infinite impulse response (IIR) approximation method called Balanced Model Truncation (BMT), we obtained a low order IIR noise control filter suitable for low-power digital signal processing system like headphones/earphones. For further performance improvement, we utilized frequency warping method so that we could obtain more accurately approximated IIR filter and we ensured system stability by reconstructing the low order IIR filter in form of cascaded second order IIR filters. ANC simulation with white noise and stability test verified that the proposed algorithm had superior attenuation performance and better robustness compared to the conventional algorithm.

Design of a Low Power Digital Filter Using Variable Canonic Signed Digit Coefficients (가변 CSD 계수를 이용한 저전력 디지털 필터의 설계)

  • Kim, Yeong-U;Yu, Jae-Taek;Kim, Su-Won
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.38 no.7
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    • pp.455-463
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    • 2001
  • In this Paper, an approximate processing method is proposed and tested. The proposed method uses variable CSD (VCSD) coefficients which approximate filter stopband attenuation by controlling the precision of the CSD coefficient sets. A decimation filter for Audio Codec '97 specifications has been designed having processor architecture that consists of program/data memory, arithmetic unit, energy/level decision, and sinc filter blocks, and fabricated with 0.6${\mu}{\textrm}{m}$ CMOS sea-of-gate technology. For the combined two halfband FIR filters in decimation filter, the number of addition operations were reduced to 63.5%, 35.7%, and 13.9%, compared to worst-case which is not an adaptive one. Experimental results show that the total power reduction rate of the filter is varying from 3.8 % to 9.0 % with respect to worst-case. The proposed approximate processing method using variable CSD coefficients is readily applicable to various kinds of filters and suitable, especially, for the speech and audio applications, like oversampling ADCs and DACs, filter banks, voice/audio codecs, etc.

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The Effect of Internal Row on Marine Riser Dynamics (Riser의 내부유체 흐름이 Riser 동적반응에 미치는 영향)

  • Hong, Nam-Seeg
    • Journal of Korean Society of Coastal and Ocean Engineers
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    • v.7 no.1
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    • pp.75-90
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    • 1995
  • A mathematical model for the dynamic analysis of a riser system with the inclusion of internal flow and nonlinear effects due to large structural displacements is developed to investigate the effect of internal flow on marine riser dynamics. The riser system accounts fir the nonlinear boundary conditions and includes a steady flow inside the pipe which is modeled as an extensible or inextensible. tubular beam subject to nonlinear three dimensional hydrodynamic loads such as current or wave excitation. Galerkin's finite element approximation and time incremental operator are implemented to derive the matrix equation of equilibrium for the finite element system and the extensibility or inextensibility condition is used to reduce degree of freedom of the system and the required computational time in the case of a nonlinear model. The algorithm is implemented to develop computer programs used in several numerical applications. The investigations of the effect of infernal flow on riser vibration due to current or wave loading are performed according to the change of various parameters such as top tension, internal flow velocity, current velocity, wave period, and so on. It is found that the effect of internal flow can be controlled by the increase of top tension. However, careful consideration has to be given in the design point particularly for the long riser under the harmonic loading such as waves. And it is also found that the consideration of nonlinear effects due to large structural displacements increases the effect of internal flow on riser dynamics.

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Consideration of Time Lag of Sea Surface Temperature due to Extreme Cold Wave - West Sea, South Sea - (한파에 따른 표층수온의 지연시간 고찰 - 서해, 남해 -)

  • Kim, Ju-Yeon;Park, Myung-Hee;Lee, Joon-Soo;Ahn, Ji-Suk;Han, In-Seong;Kwon, Mi-Ok;Song, Ji-Yeong
    • Journal of the Korean Society of Marine Environment & Safety
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    • v.27 no.6
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    • pp.701-707
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    • 2021
  • In this study, we examined the sea surface temperature (SST), air temperature (AT), and their time lag in response to an extreme cold wave in 2018 and a weak cold wave in 2019, cross-correlating these to the northern wind direction frequency. The data used in this study include SST observations of seven ocean buoys Real-time Information System for Aquaculture Environment provided by the National Institute of Fisheries Science and automatic weather station AT near them recorded every hour; null data was interpolated. A finite impulse response filter was used to identify the appropriate data period. In the extreme cold wave in 2018, the seven locations indicated low SST caused by moving cold air through the northern wind direction. A warm cold wave in 2019, the locations showed that the AT data was similar to the normal AT data, but the SST data did not change notably. During the extreme cold wave of 2018, data showed a high correlation coefficient of about 0.7 and a time lag of about 14 hours between AT and SST; during the weak cold wave of 2019, the correlation coefficient was 0.44-0.67 and time lag about 20 hours between AT and SST. This research will contribute to rapid response to such climate phenomena while minimizing aquaculture damage.