• 제목/요약/키워드: Fast Fourier Transform Algorithm

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Spectral Analysis of the ECG Using the Improved ARMA FTF Algorithm (개선된 ARMA FTF 알고리즘을 이용한 ECG 신호의 스펙트럼 해석)

  • Nam, Hyeon-Do;An, Dong-Jun;Lee, Cheol-Hui
    • Journal of Biomedical Engineering Research
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    • v.15 no.4
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    • pp.395-400
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    • 1994
  • High resolution spectral analysis is essential for ECG anaysis. The fast Fourier transform has been widely used for frequency analysis of ECG signals but this procedure provides poor resolution when the data record is short and shows Gibb's phenomena. The ARMA FTF (Fast Transversal Filter) algorithm is used for high resolution spectral analysis. The reason of unsalability of this algorithm is investigated and the method for improving the numerical stability is proposed. The proposed algorithm is applied to spectral analysis of the ECG. Since this result has less variations than the FFT based results, it can be used for the computerized diagonosis of the ECG.

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A Study on the Enhanced Time Domain Aliasing Cancellation Transform of the AC-3 Algorithm (AC-3오디오 알고리듬의 시간축 영역 에일리어징 제거 변환부 성능향상에 관한 연구)

  • 김준성;강현철;변윤식
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.2
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    • pp.13-18
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    • 2000
  • This paper presents the result of a technique to enhance TDAC in the AC-3 algorithm. To reduce block boundary noise without decreasing the performance of transform coding, We propose new special windows which improve the defect of the AC-3 algorithm that could not properly cancel aliasing in the transient period. In addition, a fast MDCT calculation algorithm based on a fast Fourier transform, is adopted.

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A Fast IFFT Algorithm for IMDCT of AAC Decoder (AAC 디코더의 IMDCT를 위한 고속 IFFT 알고리즘)

  • Chi, Hua-Jun;Kim, Tae-Hoon;Park, Ju-Sung
    • The Journal of the Acoustical Society of Korea
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    • v.26 no.5
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    • pp.214-219
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    • 2007
  • This paper proposes a new IFFT(Inverse Fast Fourier Transform) algorithm, which is proper for IMDCT(Inverse Modified Discrete Cosine Transform) of MPEG-2 AAC(Advanced Audio Coding) decoder. The $2^n$(N-point) type IMDCT is the most powerful among many IMDCT algorithms, however it includes IFFT that requires many calculation cycles. The IFFT used in $2^n$(N-point) type IMDCT employ the bit-reverse data arrangement of inputs and N/4-point complex IFFT to reduce the calculation cycles. We devised a new data arrangement method of IFFT input and $N/4^{n+1}$-type IFFT and thus we can reduce multiplication cycles, addition cycles, and ROM size.

A Study on the Dynamic Window Switching MDCT for Enhanced AC-3 Audio Filterbank (다이나믹 윈도우 스위칭기법을 적용한 AC-3 오디오 필터뱅크의 성능향상에 관한 연구)

  • 김준성
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1998.06e
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    • pp.23-26
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    • 1998
  • This paper presents a technique to enhance TDAC in the AC-3 algorithm. To reduce block boundary noise without decreasing the performance of transform coding, new special window adopted. They improves the defect of the AC-3 algorithm that could not properly cancel aliasing in the tansient period. In addition, a fast MDCT calculation algorithm based on a fast Fourier Transform, is adopted.

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Improvement of image processing speed of the 2D Fast Complex Hadamard Transform

  • Fujita, Yasuhito;Tanaka, Ken-Ichi
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2009.01a
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    • pp.498-503
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    • 2009
  • As for Hadamard Transform, because the calculation time of this transform is slower than Discrete Cosine Transform (DCT) and Fast Fourier Transform (FFT), the effectiveness and the practicality are insufficient. Then, the computational complexity can be decreased by using the butterfly operation as well as FFT. We composed calculation time of FFT with that of Fast Complex Hadamard Transform by constructing the algorithm of Fast Complex Hadamard Transform. They are indirect conversions using program of complex number calculation, and immediate calculations. We compared calculation time of them with that of FFT. As a result, the reducing the calculation time of the Complex Hadamard Transform is achieved. As for the computational complexity and calculation time, the result that quadrinomial Fast Complex Hadamard Transform that don't use program of complex number calculation decrease more than FFT was obtained.

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2048-point Low-Complexity Pipelined FFT Processor based on Dynamic Scaling (동적 스케일링에 기반한 낮은 복잡도의 2048 포인트 파이프라인 FFT 프로세서)

  • Kim, Ji-Hoon
    • Journal of IKEEE
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    • v.25 no.4
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    • pp.697-702
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    • 2021
  • Fast Fourier Transform (FFT) is a major signal processing block being widely used. For long-point FFT processing, usually more than 1024 points, its low-complexity implementation becomes very important while retaining high SQNR (Signal-to-Quantization Noise Ratio). In this paper, we present a low-complexity FFT algorithm with a simple dynamic scaling scheme. For the 2048-point pipelined FFT processing, we can reduce the number of general multipliers by half compared to the well-known radix-2 algorithm. Also, the table size for twiddle factors is reduced to 35% and 53% compared to the radix-2 and radix-22 algorithms respectively, while achieving SQNR of more than 55dB without increasing the internal wordlength progressively.

Two dimensional FFT by Polynomial Transform (Polynomial 변환을 이용한 고속 2 차원 FFT)

  • 최환석;김원하;한승수
    • Proceedings of the IEEK Conference
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    • 2003.11a
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    • pp.473-476
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    • 2003
  • We suggest 2 dimensional Fast Fourier Transform using Polynomial Transform and integer Fast Fourier Transform. Unlike conventional 2D-FFT using the direct quantization of twiddle factor, the suggested 2D-FFT adopts implemented by the lifting so that the suggested 2D-FFT is power adaptable and reversible. Since the suggested FFT performg integer-to-integer mapping, the transform can be implemented by only bit shifts and auditions without multiplications. In addition. polynomial transform severely reduces the multiplications of 2D-FFT. While preserving the reversibility, complexity of this algorithm is shown to be much lower than that of any other algorithms in terms of the numbers of additions and shifts.

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Estimation of Total Sound Pressure Level for Friction Noise Regarding a Driving Vehicle using the Extended Kalman Filter Algorithm (확장형 칼만필터 알고리즘을 활용한 차량 주행에 따른 마찰소음의 총 음압레벨 예측)

  • Dowan, Kim;Beomsoo, Han;Sungho, Mun;Deok-Soon, An
    • International Journal of Highway Engineering
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    • v.16 no.5
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    • pp.59-66
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    • 2014
  • PURPOSES : This study is to predict the Sound Pressure Level(SPL) obtained from the Noble Close ProXimity(NCPX) method by using the Extended Kalman Filter Algorithm employing the taylor series and Linear Regression Analysis based on the least square method. The objective of utilizing EKF Algorithm is to consider stochastically the effect of error because the Regression analysis is not the method for the statical approach. METHODS : For measuring the friction noise between the surface and vehicle's tire, NCPX method was used. With NCPX method, SPL can be obtained using the frequency analysis such as Discrete Fourier Transform(DFT), Fast Fourier Transform(FFT) and Constant Percentage Bandwidth(CPB) Analysis. In this research, CPB analysis was only conducted for deriving A-weighted SPL from the sound power level in terms of frequencies. EKF Algorithm and Regression analysis were performed for estimating the SPL regarding the vehicle velocities. RESULTS : The study has shown that the results related to the coefficient of determination and RMSE from EKF Algorithm have been improved by comparing to Regression analysis. CONCLUSIONS : The more the vehicle is fast, the more the SPL must be high. But in the results of EKF Algorithm, SPLs are irregular. The reason of that is the EKF algorithm can be reflected by the error covariance from the measurements.

Design of the fast adaptive digital filter for canceling the noise in the frequency domain (주파수 영역에서 잡음 제거를 위한 고속 적응 디지털 필터 설계)

  • 이재경;윤달환
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.41 no.3
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    • pp.231-238
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    • 2004
  • This paper presents the high speed noise reduction processing system using the modified discrete fourier transform(MDFT) on the frequency domain. The proposed filter uses the linear prediction coefficients of the adaptive line enhance(ALE) method based on the Sign algorithm The signals with a random noise tracking performance are examined through computer simulations. It is confirmed that the fast adaptive digital filter is realized by the high speed adaptive noise reduction(HANR) algorithm with rapid convergence on the frequency domain(FD).

Thermal Analysis of Automotive Disc Brake Using FFT-FEM (FFT-FEM을 이용한 자동차용 디스크 브레이크의 열 해석)

  • Choi, Ji-Hoon;Kim, Do-Hyung;Lee, In
    • Transactions of the Korean Society of Mechanical Engineers A
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    • v.25 no.8
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    • pp.1253-1260
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    • 2001
  • Transient thermal analysis of a three-dimensional axisymmetric automotive disk brake is presented in this paper. Temperature fields are obtained using a hybrid FFT-FEM scheme that combines Fourier transform techniques and finite element method. The use of a fast Fourier transform algorithm can avoid singularity problems and lead to inexpensive computing time. The transformed problem is solved with finite element scheme for each frequency domain. Inverse transforms are then performed for time domain solution. Numerical examples are presented for validation tests. Comparisons with analytical results show very good agreement. Also, a 3-D simulation, based upon an automotive brake disk model is performed.