[ Pobierz całość w formacie PDF ]
.R.(1980).A General-Purpose Digital Synthesizer.J.ofthe Audio Engineering Society, 28(3):106 113.[Sandy and Parker, 1982] Sandy, F.and Parker, J.(1982).Digital voice processor con-sortium interim report.Technical Report Report MTR-81w00159-01, The MITRECorp.[Sauvagerd, 1988] Sauvagerd, U.(1988).Bit-rate reduction for high quality audiosignals using floating-point wave digital filters.In Proc.of the ISCAS, pages 2031 2034.[Savioja et al., 1995] Savioja, L., Backman, J., Jrvinen, A., and Takala, T.(1995).Waveguide mesh method for low-frequency simulation of room acoustics.Proc.15th Int.Conf.Acoustics (ICA-95), Trondheim, Norway, pages 637 640.[Scavone and Smith, 1997] Scavone, G.and Smith, J.O.(1997).Digital waveguidemodeling of woodwind toneholes.In Proc.1997 Int.Computer Music Conf., Greece.Computer Music Association.[Scavone, 1997] Scavone, G.P.(1997).An Acoustic Analysis of Single-Reed Wood-wind Instruments with an Emphasis on Design and Performance Issues and DigitalWaveguide Modeling Techniques.PhD thesis, Music Dept., Stanford University.Available as CCRMA Technical Report No.STAN M 100 or from ftp://ccrma-ftp.stanford.edu/pub/Publications/Theses/GaryScavoneThesis/.[Schafer et al., 1981] Schafer, R.W., Mersereau, R.M., and Richards, M.A.(1981).Constrained iterative restoration algorithms.Proc.IEEE, 69:432 450.[Scharf, 1964] Scharf, B.(1964).Partial masking.Acustica, 14:16 23.[Scharf, 1970] Scharf, B.(1970).Critical Bands.In Tobias, J., editor, Foundations ofModern Auditory Theory, pages 159 202.Academic. REFERENCES 521[Scharf and Buus, 1986] Scharf, B.and Buus, S.(1986).Stimulus, physiology, thresh-olds.In K.R.Boff, L.K.and Thomas, J., editors, Handbook of Perception andHuman Performance, chapter 14, Basic sensory processes II.Wiley, New York.[Scharf and Houtsma, 1986] Scharf, B.and Houtsma, A.J.M.(1986).Loudness,pitch, localization, aural distortion, pathology.In K.R.Boff, L.K.and Thomas,J., editors, Handbook of Perception and Human Performance, chapter 15, Basicsensory processes II.Wiley, New York.[Schroeder and Voessing, 1986] Schroeder, E.F.and Voessing, W.(1986).High qual-ity digital audio encoding with 3.0 bits/sample using adaptive transform coding.InProc.of the 80th.AES-Convention.Preprint 2321.[Schroeder, 1970a] Schroeder, M.(1970a).Synthesis of low-peak-factor signals andbinary sequences with low autocorrelation.IEEE Trans.Information Theory, IT-16:85 89.[Schroeder, 1986] Schroeder, M.(1986).Number Theory in Science and Communi-cation.Springer Verlag, New York, NY, 2nd enlarged edition.[Schroeder et al., 1967] Schroeder, M., Flanagan, J., and Lundry, E.(1967).Band-width compression of speech by analytic-signal rooting.Proc.IEEE, 55:396 401.[Schroeder, 1954] Schroeder, M.R.(1954). Die Statistischen Parameter der Fre-quenzkurven von Grossen Raumen .Acustica, 4:594 600.See Schroeder (1987)for English translation.[Schroeder, 1962] Schroeder, M.R.(1962). Natural Sounding Artificial Reverbera-tion .J.Audio Eng.Soc., 10(3).[Schroeder, 1965] Schroeder, M.R.(1965). New method of measuring reverberationtime .J.Acoust.Soc.Am., 37:409 412.[Schroeder, 1970b] Schroeder, M.R.(1970b). Digital simulation of sound transmis-sion in reverberant spaces .J.Acoust.Soc.Am., 47(2):424 431.[Schroeder, 1987] Schroeder, M.R.(1987). Statistical parameters of the frequencyresponse curves of large rooms .J.Audio Eng.Soc., 35(5):299 306.Englishtranslation of Schroeder (1954).[Schroeder et al., 1979] Schroeder, M.R., Atal, B.S., and Hall, J.L.(1979).Opti-mizing digital speech coders by exploiting masking properties of the human ear.J.Acoust.Soc.Am., 66:1647 1652.[Schroeder and Kuttruff, 1962] Schroeder, M.R.and Kuttruff, K.H.(1962). Onfrequency response curves in rooms.Comparison of experimental, theoretical, and 522 APPLICATIONS OF DSP TO AUDIO AND ACOUSTICSMonte Carlo results for the average frequency spacing between maxima .J.Acoust.Soc.Am., 34(1):76 80.[Schroeter and Sondhi, 1994] Schroeter, J.and Sondhi, M.M.(1994).Techniquesfor estimating vocal-tract shapes from the speech signal.IEEE Trans.Speech andAudio Processing, 2(1):133 150.[Schuller, 1995] Schuller, G.(1995).A low delay filter bank for audio coding withreduced pre-echoes.In Proc.of the 99th.AES-Convention.Preprint 4088.[Schwander and Levitt, 1987] Schwander, T.and Levitt, H.(1987).Effect of two-microphone noise reduction on speech recognition by normal-hearing listeners.J.Rehab.Res.and Devel., 24:87 92.[Scott and Gerber, 1972] Scott, R.and Gerber, S.(1972).Pitch-synchronous time-compression of speech.Proceedings of the Conference for Speech CommunicationProcessing, pages 63 65.[Segelken et al., 1992] Segelken, J.M., Wu, L.J., Lau, M.Y., Tai, K.L., Shively, R.R.,and Grau, T.G.(1992).Ultra-Dense: An MCM-Based 3-D Digital Signal Processor.IEEE Trans.on Components, Hybrids and Manufacturing Tech., 15(4):438 443.[Segiguchi et al., 1983] Segiguchi, K., Ishizaka, K., Matsudaira, T.K., and Nakajima,N.(1983).A New Approach to High-Speed Digital Signal Processing Based onMicroprogramming.J.AES, 31(7):517 522.[Seitzer et al., 1988] Seitzer, D., Brandenburg, K., Kapust, R., Eberlein, E., Gerhuser,H., Popp, H., and Schott, H.(1988).Real-time implementation of low complexityadaptive transform coding.In Proc of the 84th.AES-Convention.Preprint 2581.[Seligson, 1970] Seligson, C.(1970).Comment on high-resolution frequency-wavenumber spectrum analysis.Proc.IEEE, 58:947 949.[Sellick et al., 1982] Sellick, P., Patuzzi, R., and Johnstone, B.(1982).Measurementof basilar membrane motion in the guinea pig using the Mossbauer technique.J.Acoust.Soc.Am., 72:131 141.[Seneff, 1982] Seneff, S.(1982).System to independently modify excitation and/orspectrum of speech waveform without explicit pitch extraction.IEEE Trans.Acoust.,Speech, Signal Processing, ASSP-24:358 365.[Serra, 1989] Serra, X.(1989).A System for Sound Analysis/Transformation/SynthesisBased on a Deterministic Plus Stochastic Decomposition.PhD thesis, CCRMA,Department of Music, Stanford University. REFERENCES 523[Serra and Smith, 1989] Serra, X.and Smith, J.(1989).Spectral modeling synthesis:A sound analysis/synthesis system based on a deterministic plus stochastic decom-position.In Proc.of Int.Computer Music Conf., pages 281 284, San Francisco,CA.[Serra and Smith, 1990] Serra, X.and Smith, J.(1990).Spectral modeling synthe-sis: A sound analysis/synthesis system based on a deterministic plus stochasticdecomposition.Computer Music J., 14(4):12 24.[Shan et al., 1985] Shan, T.-J., Wax, M., and Kailath, T.(1985).On spatial smoothingfor direction-of-arrival estimation of coherent signals.IEEE Trans.Acoust.Speechand Sig.Proc., ASSP-33:806 811.[Shaw, 1974] Shaw, E.(1974).The external ear.In Keidel, W.and Neff, W., editors,Handbook of Sensory Physiology, pages 455 490.Springer.[Sheingold, 1986] Sheingold, D.H., editor (1986).Analog-digital conversion hand-book, 3rd Ed.Prentice-Hall.[Shelton, 1989] Shelton, T.(1989).Synchronization in Digital Audio.In Pohlmann,K., editor, Audio in Digital Times, pages 241 250.Audio Engineering Society.[Shields, 1970] Shields, V.(1970).Separation of additive speech signals by digitalcomb filtering.Master s thesis, Department of Electrical Engineering and ComputerScience, Massachusetts Institute of Technology.[Sigelman and Preves, 1987] Sigelman, J.and Preves, D.(1987).Field trials of a newadaptive signal processor hearing aid circuit.Hearing J., 40:24 27.[Simar et al., 1992] Simar, R., Koeppen, P., Leach, J., Marshall, S., Francis, D.,Mekras, G., and Rosenstrauch, J.(1992).Floating-Point Processors Join Forcesin Parallel Processing Architectures.IEEE Micro, 12(4):60 69.[Simpson et al., 1990] Simpson, A., Moore, B., and Glasberg, B.(1990)
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