[ Pobierz całość w formacie PDF ]
.[Deutsch, 1982] Deutsch, D., editor (1982).The Psychology of Music.AP series incognition and perception.Academic Press.[Dillier et al., 1993] Dillier, N., Frölich, T., Kompis, M., Bögli, H., and Lai, W.(1993).Digital signal processing (DSP) applications for multiband loudness correctiondigital hearing aids and cochlear implants.J.Rehab.Res.and Devel., 30:95 109.[Dillon, 1985] Dillon, H.(1985).Earmolds and high frequency response modification.Hear.Instr., 36:8 12.[Dillon and Lovegrove, 1993] Dillon, H.and Lovegrove, R.(1993).Single-microphone noise reduction systems for hearing aids: A review and an evaluation.In Studebaker, G.and Hochberg, I., editors, Acoustical Factors Affecting HearingAid Performance, pages 353 372.Allyn and Bacon.[Dimino and Parladori, 1995] Dimino, G.and Parladori, G.(1995).Entropy reductionin high quality audio coding.In Proc.of the 99th.AES-Convention.Preprint 4064.[Doblinger, 1982] Doblinger, G.(1982). Optimum filter for speech enhancementusing integrated digital signal processors.Proc.1982 IEEE Int.Conf.on Acoust.Speech and Sig.Proc., pages 168 171.[Dolby, 1967] Dolby, R.M.(1967).An audio noise reduction system.J.Audio Eng.Soc., 15(4):383 388.[Dolson, 1986] Dolson, M.(1986).The phase vocoder: A tutorial.Computer MusicJournal, 10(4):14 27.REFERENCES481[Duifhuis, 1980] Duifhuis, H.(1980).Level effects in psychophysical two-tone sup-pression.J.Acoust.Soc.Am., 67:914 927.[Durbin, 1959] Durbin, J.(1959).Efficient estimation of parameters in moving-average models.Biometrika, 46:306 316.[Durlach et al., 1986] Durlach, N.I., Braida, L.D., and Ito, Y.(1986).Towards amodel for discrimination of broadband signals.J.Acoust.Soc.Am., 80:63 72.[Dyrlund and Bisgaard, 1991] Dyrlund, O.and Bisgaard, N.(1991).Acoustic feed-back margin improvements in hearing instruments using a prototype DFS (digitalfeedback suppression) system.Scand.Audiol., 20:49 53.[Eastty et al., 1995] Eastty, P., Cooke, C., Densham, R., Konishu, T., and Matsushige,T.(1995).The Hardware Behind a Large Digital Mixer.In AES 99th convention.Preprint 4124.[Edler, 1988] Edler, B.(1988).Prädiktive Teilbandcodierung mit Vektorquantisierungfür Audiosignale hoher Tonqualität (in German).In ITG Fachbericht, volume 106,pages 223 228.[Edler, 1989] Edler, B.(1989).Coding of audio signals with overlapping block trans-form and adaptive window functions.Frequenz , 43:252 256.(in German).[Edler, 1992] Edler, B.(1992).Aliasing reduction in sub-bands of cascaded filterbanks with decimation.Electronics Letters, 28:1104 1105.[Edler, 1995] Edler, B.(1995).Äquivalenz von Transformation und Teilbandzerlegungin der Quellencodierung.Dissertation, Universität Hannover.(in German).[Efron and Jeen, 1992] Efron, A.and Jeen, H.(1992).Pre-whitening for detection incorrelated plus impulsive noise.Proc.IEEE Int.Conf.Acoust., Speech and SignalProc, II:469 472.[Egan and Hake, 1950] Egan, J.and Hake, H.(1950).On the masking pattern of asimple auditory stimulus.J.Acoust.Soc.Am., 22:622 630.[Egolf, 1982] Egolf, D.(1982).Review of the acoustic feedback literature from acontrol theory point of view.In The Vanderbilt Hearing-Aid Report, Monographsin Contemporary Audiology, pages 94 103.[Egolf et al., 1986] Egolf, D., Haley, B., and Larson, V.(1986).The constant-velocitynature of hearing aids: Conclusions based on computer simulations.J.Acoust.Soc.Am., 79:1592 1602.APPLICATIONS OF DSP TO AUDIO AND ACOUSTICS482[Egolf et al., 1985] Egolf, D., Howell, H., Weaver, K., and Barker, S.(1985).Thehearing aid feedback path: Mathematical simulations, experimental verification.J.Acoust.Soc.Am., 78:1578 1587.[Egolf et al., 1978] Egolf, D., Tree, D., and Feth, L.(1978).Mathematical predictionsof electroacoustic frequency response of in situ hearing aids.J.Acoust.Soc.Am.,63:264 271.[Ellis, 1992] Ellis, D.(1992).A perceptual representation of sound.Master s the-sis, Department of Electrical Engineering and Computer Science, MassachusettsInstitute of Technology.[Ellis et al., 1991] Ellis, D., Vercoe, B., , and Quatieri, T.(1991).A perceptual repre-sentation of audio for co-channel source separation.In Proc.IEEE Workshop Appl.of Signal Processing to Audio and Acoustics, Mohonk Mountain House, New Paltz,NY.[Engebretson and French-St.George, 1993] Engebretson, A.and French-St.George,M.(1993).Properties of an adaptive feedback equalization algorithm.J.Rehab.Res.and Devel., 30:8 16.[Engebretson et al., 1990] Engebretson, A., O Connell, M., and Gong, F.(1990).Anadaptive feedback equalization algorithm for the CID digital hearing aid.Proc.12thAnnual Int.Conf.of the IEEE Eng.in Medicine, Biology Soc., Part 5:2286 2287.[Ephraim, 1992] Ephraim, Y.(1992).Statistical-model-based speech enhancementsystems.Proc.IEEE, 80(10):1526 1555.[Ephraim and Malah, 1983] Ephraim, Y.and Malah, D.(1983).Speech enhancementusing optimal non-linear spectral amplitude estimation.In Proc.IEEE Int.Conf.Acoust., Speech, Signal Processing, pages 1118 1121, Boston.[Ephraim and Malah, 1984] Ephraim, Y.and Malah, D.(1984).Speech enhancementusing a minimum mean-square error short-time spectral amplitude estimator.IEEETrans.Acoust., Speech, Signal Processing, 32(6):1109 1121.[Ephraim and Malah, 1985] Ephraim, Y.and Malah, D.(1985).Speech enhancementusing a minimum mean-square error log-spectral amplitude estimator.IEEE Trans.Acoust., Speech, Signal Processing, 33(2):443 445.[Ephraim and Van Trees, 1995] Ephraim, Y.and Van Trees, H.L.(1995).A sig-nal subspace approach for speech enhancement.IEEE Trans.Speech and AudioProcessing, 3(4):251 266.REFERENCES 483[Ephraim and VanTrees, 1993] Ephraim, Y.and VanTrees, H.(1993).A signal sub-space approach for speech enhancement.Proc.IEEE Int.Conf.Acoust., Speech andSignal Proc, II:359 362.[Erwood and Xydeas, 1990] Erwood, A.and Xydeas, C.(1990).A multiframe spectralweighting system for the enhancement of speech signals corrupted by acousticnoise.In SIGNAL PROCESSING V: Theories and Applications, pages 1107 1110.Elsevier.[Esteban and Galand, 1977] Esteban, D.and Galand, C.(1977).Application ofQuadrature Mirror Filters to Split Band Voice Coding Schemes.In Proc.IEEEInt.Conf.Acoust., Speech and Signal Proc, pages 191 195.[ETSI91tm74, 1991] ETSI91tm74 (October 1991).Global analysis of selection tests:Basic data.ETSI/TM/TM5/TCH-HS.Technical Document 91/74.[ETSIstdR06, 1992] ETSIstdR06 (1992).Speech codec specifications.ETSI/GSM.ETS R.06.[Etter and Moschytz, 1994] Etter, W.and Moschytz, G.S.(1994).Noise reduction bynoise-adaptive spectral magnitude expansion.J.Audio Eng.Soc., 42(5):341 349.[Evans et al., 1981] Evans, J., Johnson, J., and Sun, D.(1981).High resolution angularspectrum estimation techniques for terrain scattering analysis and angle of arrivalestimation.Proc.1st ASSP Workshop on Spectral Estimation, pages 5.3.1 5.3.10.[Fabry, 1991] Fabry, D.(1991).Programmable and automatic noise reduction inexisting hearing aids.In Studebaker, B.and Beck, editors, The Vanderbilt HearingAid Report II, pages 65 78.York Press.[Fabry and Tasell, 1990] Fabry, D.and Tasell, D.V.(1990).Evaluation of anarticulation-index based model for predicting the effects of adaptive frequencyresponse hearing aids.J.Speech and Hearing Res., 33:676 689.[Fairbanks et al., 1954] Fairbanks, G., Everitt, W., and Jaeger, R.(1954).Methodfor time or frequency compression-expansion of speech.IEEE Trans.Audio andElectroacoustics , AU-2:7 12.[Ferrara, 1980] Ferrara, E.(1980).Fast implementation of LMS adaptive filters.IEEETrans.Acoust.Speech and Sig.Proc., Vol ASSP-28:474 475.[Fettweis, 1986] Fettweis, A.(1986).Wave digital filters: Theory and practice.Proc.IEEE, 74(2):270 327.[Fielder, 1985] Fielder, L.D.(1985).Modulation Noise in Floating-Point ConversionSystems.J
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