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Volumn , Issue , 2009, Pages 456-463

Audio FFT filter banks

Author keywords

[No Author keywords available]

Indexed keywords

ECONOMIC AND SOCIAL EFFECTS; FILTER BANKS;

EID: 84872710211     PISSN: 24136700     EISSN: 24136689     Source Type: Conference Proceeding    
DOI: None     Document Type: Conference Paper
Times cited : (11)

References (24)
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    • Stevens, S.S.1    Davis, H.2
  • 6
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    • Calculation of a constant Q spectral transform
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    • Brown, J.C.1
  • 7
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    • An efficient algorithm for the calculation of a constant Q transform
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    • (1992) J. Acoust. Soc. of Amer. , vol.92 , Issue.5 , pp. 2698-2701
    • Brown, J.C.1    Puckette, M.S.2
  • 8
    • 0036565284 scopus 로고    scopus 로고
    • A model of loudness applicable to time-varying sounds
    • May
    • Brian R. Glasberg and Brian C. J. Moore, "A model of loudness applicable to time-varying sounds," J. Audio Eng. Soc., vol. 50, no. 5, pp. 331-342, May 2002.
    • (2002) J. Audio Eng. Soc. , vol.50 , Issue.5 , pp. 331-342
    • Glasberg, B.R.1    Moore, B.C.J.2
  • 14
    • 0017552006 scopus 로고
    • A unified approach to short-time Fourier analysis and synthesis
    • Nov.
    • Jont B. Allen and Larry R. Rabiner, "A unified approach to short-time Fourier analysis and synthesis," Proc. IEEE, vol. 65, no. 11, pp. 1558-1564, Nov. 1977.
    • (1977) Proc. IEEE , vol.65 , Issue.11 , pp. 1558-1564
    • Allen, J.B.1    Rabiner, L.R.2
  • 15
    • 0016962212 scopus 로고
    • Implementation of the digital phase vocoder using the fast Fourier transform
    • June
    • M. R. Portnoff, "Implementation of the digital phase vocoder using the fast Fourier transform," IEEE Trans. Acoustics, Speech, Signal Processing, vol. ASSP-24, no. 3, pp. 243-248, June 1976.
    • (1976) IEEE Trans. Acoustics, Speech, Signal Processing , vol.ASSP-24 , Issue.3 , pp. 243-248
    • Portnoff, M.R.1
  • 17
  • 19
    • 0031636309 scopus 로고    scopus 로고
    • FFTW: An adaptive software architecture for the FFT
    • Seattle, New York, IEEE Press
    • M. Frigo and S. G. Johnson, "FFTW: An adaptive software architecture for the FFT," in Proc. Int. Conf. Acoustics, Speech, and Signal Processing, Seattle, New York, 1998, vol. 3, pp. 1381-1384, IEEE Press, http://www.fftw.org/.
    • (1998) Proc. Int. Conf. Acoustics, Speech, and Signal Processing , vol.3 , pp. 1381-1384
    • Frigo, M.1    Johnson, S.G.2
  • 20
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    • Various
    • (Various), Prime-Factor FFT Algorithm, http://www.wikipedia.org/wiki/- Prime-factorFFTalgorithm, 2003.
    • (2003) Prime-Factor FFT Algorithm
  • 21
    • 0031012771 scopus 로고    scopus 로고
    • A time-domain, level-dependent auditory filter: The gammachirp
    • Toshio Irino and Roy D. Patterson, "A time-domain, level-dependent auditory filter: The gammachirp," J. Acoust. Soc. of Amer., vol. 101, pp. 412-419, 1997.
    • (1997) J. Acoust. Soc. of Amer. , vol.101 , pp. 412-419
    • Irino, T.1    Patterson, R.D.2
  • 22
    • 0000177937 scopus 로고    scopus 로고
    • The Dolph-Chebyshev window: A simple optimal filter
    • Apr.
    • Peter Lynch, "The Dolph-Chebyshev window: A simple optimal filter," Amer. Meteorological Soc. J. Online, vol. 125, no. 4, pp. 655-660, Apr. 1997.
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    • Lynch, P.1
  • 23
    • 39749202229 scopus 로고    scopus 로고
    • Recent advances and new design method in nonuniform filter banks
    • June
    • Guangming Shi, Xuemei Xie, Xuyang Chen, and Wei Zhong, "Recent advances and new design method in nonuniform filter banks," in Proc. IEEE Int. Conf. Comm., Circ. & Sys., June 2006, vol. 1, pp. 211-215.
    • (2006) Proc. IEEE Int. Conf. Comm., Circ. & Sys. , vol.1 , pp. 211-215
    • Shi, G.1    Xie, X.2    Chen, X.3    Zhong, W.4


* 이 정보는 Elsevier사의 SCOPUS DB에서 KISTI가 분석하여 추출한 것입니다.