Applications of digital signal processing to audio and acoustics

With the advent of `multimedia', digital signal processing (DSP) of sound has emerged from the shadow of bandwidth limited speech processing to become a research field of its own. To date, most research in DSP applied to sound has been concentrated on speech, which is bandwidth limited to about 4 kilohertz. Speech processing is also limited by the low fidelity typically expected in the telephone network. Today, the main applications of audio DSP are high quality audio coding and the digital generation and manipulation of music signals. They share common research topics including perceptual measurement techniques and analysis/synthesis methods. Additional important topics are hearing aids using signal processing technology and hardware architectures for digital signal processing of audio. In all these areas the last decade has seen a significant amount of application-oriented research. The frequency range of wideband audio has an upper limit of 20 kilohertz and the resulting difference in frequency range and Signal to Noise Ratio (SNR) due to sample size must be taken into account when designing DSP algorithms. There are whole classes of algorithms that the speech community is not interested in pursuing or using. These algorithms and techniques are revealed in this book. This book is suitable for advanced level courses and serves as a valuable reference for researchers in the field. Interested and informed engineers will also find the book useful in their work.

[1]  T. W. Parsons,et al.  Enhancing/Intelligibility of Speech in Noisy or Multi-Talker Environments. , 1975 .

[2]  Henrique S. Malvar Lapped transforms for efficient transform/subband coding , 1990, IEEE Trans. Acoust. Speech Signal Process..

[3]  D J Van Tasell,et al.  Robust adaptive microphone array processing for hearing aids: realistic speech enhancement. , 1994, The Journal of the Acoustical Society of America.

[4]  Dennis L. Wilson,et al.  Some improvements on the synchronized-overlap-add method of time scale modification for use in real-time speech compression and noise filtering , 1988, IEEE Trans. Acoust. Speech Signal Process..

[5]  David Malah,et al.  Time-domain algorithms for harmonic bandwidth reduction and time scaling of speech signals , 1979 .

[6]  Jelena Kovacevic,et al.  Wavelets and Subband Coding , 2013, Prentice Hall Signal Processing Series.

[7]  M. Schroeder New Method of Measuring Reverberation Time , 1965 .

[8]  B. Moore An Introduction to the Psychology of Hearing , 1977 .

[9]  E. Zwicker,et al.  Das Ohr als Nachrichtenempfänger , 1967 .

[10]  J Jerger,et al.  Speech understanding in the elderly. , 1989, Ear and hearing.

[11]  R. Lippmann,et al.  Study of multichannel amplitude compression and linear amplification for persons with sensorineural hearing loss. , 1981, The Journal of the Acoustical Society of America.

[12]  M W Skinner,et al.  Speech intelligibility in noise-induced hearing loss: effects of high-frequency compensation. , 1980, The Journal of the Acoustical Society of America.

[13]  Byeong Gi Lee,et al.  Lossy pole-zero modeling for speech signals , 1996, IEEE Trans. Speech Audio Process..

[14]  R A Lutfi,et al.  Additivity of simultaneous masking. , 1983, The Journal of the Acoustical Society of America.

[15]  Barry Truax,et al.  Discovering Inner Complexity: Time Shifting and Transposition with a Real-Time Granulation Technique , 1994 .

[16]  A. H. Marshall,et al.  Spatial impression due to early lateral reflections in concert halls: The derivation of a physical measure , 1981 .

[17]  H Levitt,et al.  Effect of two-microphone noise reduction on speech recognition by normal-hearing listeners. , 1987, Journal of rehabilitation research and development.

[18]  Thomas F. Quatieri,et al.  An approach to co-channel talker interference suppression using a sinusoidal model for speech , 1990, IEEE Trans. Acoust. Speech Signal Process..

[19]  Julius O. Smith,et al.  Performance Expression in Commuted Waveguide Synthesis of Bowed Strings , 1995, ICMC.

[20]  L D Braida,et al.  Principal-component amplitude compression for the hearing impaired. , 1987, The Journal of the Acoustical Society of America.

[21]  H. E. Roys Disc recording and reproduction , 1978 .

[22]  Alan V. Oppenheim,et al.  Evaluation of a speech enhancement system , 1977 .

[23]  Simon J. Godsill,et al.  Robust noise reduction for speech and audio signals , 1996, 1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings.

[24]  A.V. Oppenheim,et al.  The importance of phase in signals , 1980, Proceedings of the IEEE.

[25]  M. Misaki,et al.  Time-scale modification of speech signals using cross-correlation functions , 1992 .

[26]  James A. Moorer A Flexible Method for Synchronizing Parameter Updates for Real-Time Audio Signal Processors , 1985 .

[27]  Kuansan Wang,et al.  Auditory representations of acoustic signals , 1992, IEEE Trans. Inf. Theory.

[28]  F L Wightman,et al.  Headphone simulation of free-field listening. I: Stimulus synthesis. , 1989, The Journal of the Acoustical Society of America.

[29]  K A Weaver,et al.  The hearing aid feedback path: mathematical simulations and experimental verification. , 1985, The Journal of the Acoustical Society of America.

[30]  P. Morse Vibration and Sound , 1949, Nature.

[31]  Ioannis Pitas,et al.  Nonlinear Digital Filters - Principles and Applications , 1990, The Springer International Series in Engineering and Computer Science.

[32]  Jae S. Lim,et al.  Signal estimation from modified short-time Fourier transform , 1983, ICASSP.

[33]  S. V. Vaseghi,et al.  Restoration of Old Gramophone Recordings , 1992 .

[34]  A. J. Efron,et al.  Pre-whitening for detection in correlated plus impulsive noise , 1992, [Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[35]  L. H. Anauer,et al.  Speech Analysis and Synthesis by Linear Prediction of the Speech Wave , 2000 .

[36]  Jürgen Herre,et al.  MPEG-2 NBC Audio-Stereo and Multichannel Coding Methods , 1996 .

[37]  T. Ramstad,et al.  Cosine-modulated analysis-synthesis filterbank with critical sampling and perfect reconstruction , 1991, [Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing.

[38]  C V Pavlovic,et al.  Derivation of primary parameters and procedures for use in speech intelligibility predictions. , 1987, The Journal of the Acoustical Society of America.

[39]  Kaoru Arakawa,et al.  Separation of a Nonstationary Component from the EEG by a Nonlinear Digital Filter , 1986, IEEE Transactions on Biomedical Engineering.

[40]  Curtis Abbott,et al.  The Digital Audio Processing Station: A New Concept in Audio Postproduction , 1986 .

[41]  Michael J. Flynn,et al.  Introduction to Arithmetic for Digital Systems Designers , 1995 .

[42]  S. Seneff System to independently modify excitation and/Or spectrum of speech waveform without explicit pitch extraction , 1982 .

[43]  Matti Karjalainen,et al.  Improving the kelly-lochbaum vocal tract model using conical tube sections and fractional delay filtering techniques , 1994, ICSLP.

[44]  Kuldip K. Paliwal,et al.  A speech enhancement method based on Kalman filtering , 1987, ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[45]  W. K. Hastings,et al.  Monte Carlo Sampling Methods Using Markov Chains and Their Applications , 1970 .

[46]  Curtis Abbott,et al.  The 4C Machine , 1979 .

[47]  W. Jesteadt,et al.  Forward masking as a function of frequency, masker level, and signal delay. , 1982, The Journal of the Acoustical Society of America.

[48]  H Levitt,et al.  Effect of release time in compression hearing aids: paired-comparison judgments of quality. , 1995, The Journal of the Acoustical Society of America.

[49]  John Mourjopoulos,et al.  Noisy Audio Signal Enhancement Using Subjective Spectra , 1992 .

[50]  Ernst Eberlein,et al.  Combined Stereo Coding , 1992 .

[51]  Peter Vary,et al.  Noise suppression by spectral magnitude estimation —mechanism and theoretical limits— , 1985 .

[52]  Lawrence R. Rabiner,et al.  Digital Techniques for Changing the Sampling Rate of a Signal , 1982 .

[53]  Maxim J. Goldberg,et al.  Removing noise from music using local trigonometric bases and wavelet packets , 1994 .

[54]  Ulrich Heute,et al.  A new approach to objective quality-measures based on attribute-matching , 1992, Speech Commun..

[55]  W P Gibson,et al.  The Nucleus 22-channel cochlear implant system. , 1997, Advances in oto-rhino-laryngology.

[56]  S. Vernon Design and implementation of AC-3 coders , 1995 .

[57]  Simon J. Godsill,et al.  Robust treatment of impulsive noise in speech and audio signals , 1996 .

[58]  K Mizoi,et al.  Clinical results of hearing aid with noise-level-controlled selective amplification. , 1983, Audiology : official organ of the International Society of Audiology.

[59]  Deepen Sinha,et al.  Low bit rate transparent audio compression using adapted wavelets , 1993, IEEE Trans. Signal Process..

[60]  Daniel P. Siewiorek,et al.  Parallel processing: the Cm* experience , 1986 .

[61]  Douglas D. O'Shaughnessy,et al.  Statistical recovery of wideband speech from narrowband speech , 1992, IEEE Trans. Speech Audio Process..

[62]  Jens Blauert,et al.  Principles of binaural room simulation , 1992 .

[63]  J. Martens,et al.  A new theory for multitone masking. , 1982, The Journal of the Acoustical Society of America.

[64]  H Levitt,et al.  Effect of compression ratio in a slow-acting compression hearing aid: paired-comparison judgments of quality. , 1994, The Journal of the Acoustical Society of America.

[65]  William A. Wulf,et al.  HYDRA/C.Mmp, An Experimental Computer System , 1981 .

[66]  L. Humes,et al.  Models of the additivity of masking. , 1989, The Journal of the Acoustical Society of America.

[67]  T.F. Quatieri,et al.  A perceptual representation of audio for co-channel source separation , 1991, Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics.

[68]  Randy L. Haupt,et al.  Introduction to Adaptive Arrays , 1980 .

[69]  James A. Moorer The Audio Signal Processor: The Next Step in Digital Audio , 1982 .

[70]  S. Tewksbury,et al.  Terminology related to the performance of S/H, A/D, and D/A circuits , 1978 .

[71]  Rodnay Zaks,et al.  A/D and D/A conversion , 1978, Microprocess..

[72]  Julius O. Smith,et al.  The Second-Order Digital Waveguide Oscillator , 1992, ICMC.

[73]  Robert B. Dunn,et al.  Time-Scale Modification of Complex Acoustic Signals in Noise , 1994 .

[74]  D A Preves,et al.  Hearing aid saturation and aided loudness discomfort. , 1992, Journal of speech and hearing research.

[75]  A M Engebretson,et al.  Properties of an adaptive feedback equalization algorithm. , 1993, Journal of rehabilitation research and development.

[76]  Ernst F Schroeder,et al.  Aspec-Adaptive Spectral Entropy Coding of High Quality Music Signals , 1991 .

[77]  M. R. Schroeder,et al.  Digital simulation of sound transmission in reverberant spaces (part 1) , 1969 .

[78]  B. Moore,et al.  Gap detection and masking in hearing-impaired and normal-hearing subjects. , 1987, The Journal of the Acoustical Society of America.

[79]  Marina Bosi,et al.  Overview of MPEG audio : Current and future standards for low-bit-rate audio coding , 1997 .

[80]  Karlheinz Brandenburg,et al.  Near-lossless coding of high quality digital audio: first results , 1993, 1993 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[81]  D A Fabry,et al.  Evaluation of an articulation-index based model for predicting the effects of adaptive frequency response hearing aids. , 1990, Journal of speech and hearing research.

[82]  Miller S. Puckette,et al.  Phase-locked vocoder , 1995, Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics.

[83]  Thomas F. Quatieri,et al.  Pitch estimation and voicing detection based on a sinusoidal speech model , 1990, International Conference on Acoustics, Speech, and Signal Processing.

[84]  Peter Charles Eastty,et al.  The Hardware Behind a Large Digital Mixer , 1995 .

[85]  Simon J. Godsill,et al.  A two-channel approach to the removal of impulsive noise from archived recordings , 1994, Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing.

[86]  M. J. Carey,et al.  A system for reducing impulsive noise on gramophone reproduction equipment , 1980 .

[87]  Daniel W. Gravereaux,et al.  Automatic Detection of Impulse Noise , 1971 .

[88]  David Malah,et al.  Speech enhancement using optimal non-linear spectral amplitude estimation , 1983, ICASSP.

[89]  J. C. Steinberg,et al.  Factors Governing the Intelligibility of Speech Sounds , 1945 .

[90]  Ian Robertson Sinclair Audio Electronics Reference Book , 1989 .

[91]  Yrjö Neuvo,et al.  Use of short floating-point formats in audio applications , 1992 .

[92]  James A. Moorer,et al.  The Use of the Phase Vocoder in Computer Music Applications , 1976 .

[93]  Jean Kergomard,et al.  On the reflection functions associated with discontinuities in conical bores , 1990 .

[94]  R. J. McAulay,et al.  Computationally efficient sine-wave synthesis and its application to sinusoidal transform coding , 1988, ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing.

[95]  Mendel Kleiner,et al.  Auralization-An Overview , 1993 .

[96]  E. Zwicker Procedure for calculating loudnesss of temporally variable sounds. , 1977, The Journal of the Acoustical Society of America.

[97]  Ahmed H. Tewfik,et al.  Low bit rate high quality audio coding with combined harmonic and wavelet representations , 1996, 1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings.

[98]  I. J. Leontaritis,et al.  Input-output parametric models for non-linear systems Part II: stochastic non-linear systems , 1985 .

[99]  John Wawrzynek,et al.  MIMIC, A Custom VLSI Parallel Processor for Musical Sound Synthesis , 1990 .

[100]  D. A. Gray,et al.  A structured gradient algorithm for adaptive beamforming , 1989 .

[101]  S. Vaseghi Detection and suppression of impulsive noise in speech communication systems , 1990 .

[102]  W. Voessing,et al.  High Quality Digital Audio Encoding with 3.0 Bits/Sample Using Adaptive Transform Coding , 1986 .

[103]  W. T. Peake,et al.  Input impedance of the cochlea in cat. , 1982, The Journal of the Acoustical Society of America.

[104]  Robert C. Maher,et al.  A Method for Extrapolation of Missing Digital Audio Data , 1994 .

[105]  S. D. Gray,et al.  Filtering of colored noise for speech enhancement and coding , 1989, International Conference on Acoustics, Speech, and Signal Processing,.

[106]  H Levitt,et al.  Evaluation of orthogonal polynomial compression. , 1991, The Journal of the Acoustical Society of America.

[107]  Patrick M. Zurek,et al.  Reducing acoustic feedback in hearing aids , 1995, IEEE Trans. Speech Audio Process..

[108]  Jim Woodhouse,et al.  On the fundamentals of bowed string dynamics , 1979 .

[109]  Josef Schmee,et al.  Outliers in Statistical Data (2nd ed.) , 1986 .

[110]  D P Egolf,et al.  The constant-volume-velocity nature of hearing aids: conclusions based on computer simulations. , 1986, The Journal of the Acoustical Society of America.

[111]  Joseph Rothweiler,et al.  Polyphase quadrature filters-A new subband coding technique , 1983, ICASSP.

[112]  Eric Moulines,et al.  Non-parametric techniques for pitch-scale and time-scale modification of speech , 1995, Speech Commun..

[113]  D. Rubin,et al.  Maximum likelihood from incomplete data via the EM - algorithm plus discussions on the paper , 1977 .

[114]  Ernst Eberlein,et al.  Real-Time Implementation of Low Complexity Adaptive Transform Coding , 1988 .

[115]  Curtis Roads,et al.  Foundations of computer music , 1985 .

[116]  B. M. Johnstone,et al.  Measurement of basilar membrane motion in the guinea pig using the Mössbauer technique. , 1982, The Journal of the Acoustical Society of America.

[117]  John T. Scott,et al.  Fundamentals of musical acoustics , 1976 .

[118]  J. Smith,et al.  The Carathéodory-Fejér method for recursive digital filter design , 1983 .

[119]  Henry Cox,et al.  Practical supergain , 1986, IEEE Trans. Acoust. Speech Signal Process..

[120]  John G. Beerends,et al.  Modeling a Cognitive Aspect in the Measurement of the Quality of Music Codecs , 1994 .

[121]  W Soede,et al.  Assessment of a directional microphone array for hearing-impaired listeners. , 1993, The Journal of the Acoustical Society of America.

[122]  L. R. Rabiner,et al.  Improving the quality of a noisy speech signal , 1981, The Bell System Technical Journal.

[123]  Chrysostomos L. Nikias,et al.  Advanced Topics in Digital Signal Processing , 1992 .

[124]  Robert B. Dunn,et al.  A subband approach to time-scale expansion of complex acoustic signals , 1995, IEEE Trans. Speech Audio Process..

[125]  M. Portnoff,et al.  Time-scale modification of speech based on short-time Fourier analysis , 1981 .

[126]  Yariv Ephraim,et al.  A signal subspace approach for speech enhancement , 1995, IEEE Trans. Speech Audio Process..

[127]  J. E. Thornton Design of a Computer: The Control Data 6600 , 1970 .

[128]  Gabriel Weinreich,et al.  Coupled piano strings , 1977 .

[129]  J M Festen,et al.  The effect of varying the amplitude-frequency response on the masked speech-reception threshold of sentences for hearing-impaired listeners. , 1989, The Journal of the Acoustical Society of America.

[130]  B. Liu,et al.  Error bounds for jittered sampling , 1965 .

[131]  J. Cooley,et al.  The Fast Fourier Transform , 1975 .

[132]  Simon J. Godsill,et al.  A Bayesian approach to the restoration of degraded audio signals , 1995, IEEE Trans. Speech Audio Process..

[133]  John E. Markel,et al.  Linear Prediction of Speech , 1976, Communication and Cybernetics.

[134]  W. R. Bennett,et al.  Spectra of quantized signals , 1948, Bell Syst. Tech. J..

[135]  Mark J. T. Smith,et al.  Analysis-by-Synthesis/Overlap-Add Sinusoidal Modeling Applied to the Analysis and Synthesis of Musical Tones , 1992 .

[136]  David J. Goodman,et al.  Subjective quality of the same speech transmission conditions in seven different countries , 1982, ICASSP.

[137]  W. Hays,et al.  The architecture, instruction set and development support for the WE®DSP32 digital signal processor , 1986, ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[138]  Donald Geman,et al.  Stochastic relaxation, Gibbs distributions, and the Bayesian restoration of images , 1984 .

[139]  R. M. Sachs,et al.  Anthropometric manikin for acoustic research. , 1975, The Journal of the Acoustical Society of America.

[140]  D. Esteban,et al.  Application of quadrature mirror filters to split band voice coding schemes , 1977 .

[141]  J. Valiere,et al.  DÉTECTION ET SUPPRESSION DE BRUITS IMPULSIONNELS APPLIQUÉS À LA RESTAURATION D'ENREGISTREMENTS ANCIENS , 1990 .

[142]  R F Hess,et al.  Effects of flanking noise bands on the rate of growth of loudness of tones in normal and recruiting ears. , 1985, The Journal of the Acoustical Society of America.

[143]  A. Papoulis Signal Analysis , 1977 .

[144]  Thomas W. Parks,et al.  GENERATION AND COMBINATION OF GRAINS FOR MUSIC SYNTHESIS. , 1988 .

[145]  J. Pickles An Introduction to the Physiology of Hearing , 1982 .

[146]  Kevin Karplus,et al.  Digital Synthesis of Plucked-String and Drum Timbers , 1983 .

[147]  Richard Kronland-Martinet,et al.  Analysis of Sound Patterns through Wavelet transforms , 1987, Int. J. Pattern Recognit. Artif. Intell..

[148]  T. Ken Matsudaira,et al.  A New Approach to High Speed Digital Signal Processing Based on Microprogramming , 1983 .

[149]  A.V. Oppenheim,et al.  Enhancement and bandwidth compression of noisy speech , 1979, Proceedings of the IEEE.

[150]  Manfred R. Schroeder,et al.  Statistical parameters of the frequency response curves of large rooms , 1987 .

[151]  Dean Wallraff,et al.  The DMX-1000 Signal Processing Computer , 1978, ICMC.

[152]  N. Kikuma,et al.  Toeplitzization of correlation matrix in multipath environment , 1986, ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[153]  E Villchur,et al.  Signal processing to improve speech intelligibility in perceptive deafness. , 1973, The Journal of the Acoustical Society of America.

[154]  Robert B. Dunn,et al.  Time-scale modification of complex acoustic signals , 1993, 1993 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[155]  Stéphane Mallat,et al.  Singularity detection and processing with wavelets , 1992, IEEE Trans. Inf. Theory.

[156]  Alan V. Oppenheim,et al.  Evaluation of an adaptive comb filtering method for enhancing speech degraded by white noise addition , 1978 .

[157]  Henrique S. Malvar,et al.  Signal processing with lapped transforms , 1992 .

[158]  Dana Massie The Emulator II Computer Music Environment , 1985, ICMC.

[159]  Julius O. Smith,et al.  Spectral modeling synthesis: A sound analysis/synthesis based on a deterministic plus stochastic decomposition , 1990 .

[160]  Ephraim Speech enhancement using a minimum mean square error short-time spectral amplitude estimator , 1984 .

[161]  Matti Karjalainen,et al.  Modeling of Woodwind Bores with Finger Holes , 1993, ICMC.

[162]  B. Yegnanarayana,et al.  Design of recursive group-delay filters by autoregressive modeling , 1982 .

[163]  M J Penner,et al.  The coding of intensity and the interaction of forward and backward masking. , 1979, The Journal of the Acoustical Society of America.

[164]  Ajm Adrian Houtsma,et al.  Non-linear behaviour of single-reed woodwind musical instrumentx , 1991 .

[165]  S Hayashi,et al.  An objective quality assessment method for bit-reduction coding of wideband speech. , 1992, The Journal of the Acoustical Society of America.

[166]  R. R. Shively,et al.  Ultra-Dense: an MCM-based 3-D digital signal processor , 1992 .

[167]  Douglas H. Keefe,et al.  Theory of the single woodwind tone hole , 1982 .

[168]  John Princen,et al.  Audio coding with signal adaptive filterbanks , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.

[169]  Xavier Rodet,et al.  Diphone Sound Synthesis Based on Spectral Envelopes and Harmonic/Noise Excitation Functions , 1988, ICMC.

[170]  A M Simpson,et al.  Spectral enhancement to improve the intelligibility of speech in noise for hearing-impaired listeners. , 1990, Acta oto-laryngologica. Supplementum.

[171]  N. Fliege,et al.  Multirate Digital Reverberation System , 1990 .

[172]  Marc Le Brun,et al.  Digital Waveshaping Synthesis , 1979 .

[173]  Max V. Mathews,et al.  The Technology Of Computer Music , 1970 .

[174]  John Mourjopoulos,et al.  Speech enhancement using psychoacoustic criteria , 1993, 1993 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[175]  John P. Roesgen The ADSP-2100 DSP Microprocessor , 1986, IEEE Micro.

[176]  A. Gray,et al.  Unconstrained frequency-domain adaptive filter , 1982 .

[177]  George Carayannis,et al.  Speech enhancement from noise: A regenerative approach , 1991, Speech Commun..

[178]  Henrique S. Malvar Extended lapped transforms: fast algorithms and applications , 1991, [Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing.

[179]  C. K. Yuen,et al.  Theory and Application of Digital Signal Processing , 1978, IEEE Transactions on Systems, Man, and Cybernetics.

[180]  M. Lang,et al.  Simple and robust method for the design of allpass filters using least-squares phase error criterion , 1994 .

[181]  J. D. Rhodes,et al.  Explicit solution for the synthesis of two-variable transmission-line networks , 1973 .

[182]  Sergio Cavaliere,et al.  MARS: The X20 Device and SM100 Board , 1992, International Conference on Mathematics and Computing.

[183]  W. Frank,et al.  Improved vocal tract models for speech synthesis , 1986, ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[184]  R. McAulay,et al.  Speech enhancement using a soft-decision noise suppression filter , 1980 .

[185]  A. Peterson,et al.  Transform domain LMS algorithm , 1983 .

[186]  R. McAulay,et al.  "Multirate sinusoidal transform coding at rates from 2.4 kbps to 8 kbps" , 1987, ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[187]  Manfred R. Schroeder,et al.  Synthesis of low-peak-factor signals and binary sequences with low autocorrelation (Corresp.) , 1970, IEEE Trans. Inf. Theory.

[188]  Xavier Rodet,et al.  The CHANT Project: From the Synthesis of the Singing Voice to Synthesis in General , 1984 .

[189]  Davide Rocchesso,et al.  Circulant and elliptic feedback delay networks for artificial reverberation , 1997, IEEE Trans. Speech Audio Process..

[190]  Howell Tong,et al.  Non-Linear Time Series , 1990 .

[191]  Butler W. Lampson,et al.  The Memory System of a High-Performance Personal Computer , 1981, IEEE Transactions on Computers.

[192]  Luís B. Almeida,et al.  Frequency-varying sinusoidal modeling of speech , 1989, IEEE Trans. Acoust. Speech Signal Process..

[193]  G. K. Yates,et al.  Basilar membrane measurements and the travelling wave , 1986, Hearing Research.

[194]  Jean-Marc Jot,et al.  An analysis/synthesis approach to real-time artificial reverberation , 1992, [Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[195]  D Byrne,et al.  The effects of multichannel compression/expansion amplification on the intelligibility of nonsense syllables in noise. , 1984, The Journal of the Acoustical Society of America.

[196]  D. Gareth Loy Notes on the Implementation of MUSBOX: A Compiler for the Systems Concepts Digital Synthesizer , 1981 .

[197]  A. Hirschberg,et al.  Mechanics of musical instruments , 1995 .

[198]  Douglas Preis,et al.  Restoration of nonlinearly distorted magnetic recordings , 1983, ICASSP.

[199]  Fengmin Gong,et al.  An Adaptive Feedback Equalization Algorithm For The CID Digital Hearing Aid , 1990, [1990] Proceedings of the Twelfth Annual International Conference of the IEEE Engineering in Medicine and Biology Society.

[200]  M. Ross,et al.  Average magnitude difference function pitch extractor , 1974 .

[201]  M Kompis,et al.  Digital signal processing (DSP) applications for multiband loudness correction digital hearing aids and cochlear implants. , 1993, Journal of rehabilitation research and development.

[202]  Pjw Rayner,et al.  The effects of non-stationary signal characteristics on the performance of adaptive audio restoration systems , 1989, International Conference on Acoustics, Speech, and Signal Processing,.

[203]  Davide Rocchesso,et al.  Connections between Feedback Delay Networks and Waveguide Networks for Digital Reverberation , 1994, ICMC.

[204]  James A. Moorer,et al.  Linear-Phase Bandsplitting: Theory and Applications , 1986 .

[205]  R H Brey,et al.  Improvement in speech intelligibility in noise employing an adaptive filter with normal and hearing-impaired subjects. , 1987, Journal of rehabilitation research and development.

[206]  Douglas D. O'Shaughnessy,et al.  Speech enhancement based conceptually on auditory evidence , 1991, IEEE Trans. Signal Process..

[207]  Robert B. Dunn,et al.  Underwater Signal Enhancement Using A Sine-wave Representation , 1992, OCEANS 92 Proceedings@m_Mastering the Oceans Through Technology.

[208]  Tracy Petersen,et al.  Acoustic noise suppression in the context of a perceptual model , 1981, ICASSP.

[209]  C. W. Therrien,et al.  Methods for acoustic data synthesis , 1994, Proceedings of IEEE 6th Digital Signal Processing Workshop.

[210]  Dave Francis,et al.  Floating-point processors join forces in parallel processing architectures , 1992, IEEE Micro.

[211]  A Boothroyd,et al.  Amplitude compression and profound hearing loss. , 1988, Journal of speech and hearing research.

[212]  B C Moore,et al.  A comparison of behind-the-ear high-fidelity linear hearing aids and two-channel compression aids, in the laboratory and in everyday life. , 1983, British journal of audiology.

[213]  Luís B. Almeida,et al.  Quasi-optimal analysis for sinusoidal representation of speech , 1987 .

[214]  Wj Fitzgerald,et al.  Interpolation of missing samples for audio restoration , 1994 .

[215]  J. Flanagan,et al.  Signal models for low bit‐rate coding of speech , 1980 .

[216]  Henry Cox,et al.  Robust adaptive beamforming , 2005, IEEE Trans. Acoust. Speech Signal Process..

[217]  R. McDonough Degraded Performance of Nonlinear Array Processors in the Presence of Data Modeling Errors , 1972 .

[218]  R M Shiffrin,et al.  Nonlinearities in the coding of intensity within the context of a temporal summation model. , 1980, The Journal of the Acoustical Society of America.

[219]  M.G. Bellanger,et al.  Digital processing of speech signals , 1980, Proceedings of the IEEE.

[220]  A. Dale Magoun,et al.  Decision, estimation and classification , 1989 .

[221]  J. L. Goldstein,et al.  A central spectrum model: a synthesis of auditory-nerve timing and place cues in monaural communication of frequency spectrum. , 1983, The Journal of the Acoustical Society of America.

[222]  Simon J. Godsill,et al.  Bayesian Enhancement of Speech and Audio Signals which can be Modelled as ARMA Processes , 1997 .

[223]  Eric Lindemann DSP Architectures for the Digital Audio Workstation , 1987 .

[224]  Guy R. L. Sohie,et al.  A digital signal processor with IEEE floating-point arithmetic , 1988, IEEE Micro.

[225]  Jean-Baptiste Barrière,et al.  A Digital Signal Multiprocessor and its Musical Application , 1989, ICMC.

[226]  A C Neuman,et al.  The effect of filtering on the intelligibility and quality of speech in noise. , 1987, Journal of rehabilitation research and development.

[227]  D P Egolf,et al.  Mathematical predictions of electroacoustic frequency response of in situ hearing aids. , 1978, The Journal of the Acoustical Society of America.

[228]  J M Kates,et al.  Quality ratings for frequency-shaped peak-clipped speech. , 1994, The Journal of the Acoustical Society of America.

[229]  Raymond N. J. Veldhuis,et al.  Adaptive interpolation of discrete-time signals that can be modeled as autoregressive processes , 1986, IEEE Trans. Acoust. Speech Signal Process..

[230]  R. J. McAulay,et al.  Speech transformations based on a sinusoidal representation , 1985, ICASSP '85. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[231]  Werner Verhelst,et al.  An overlap-add technique based on waveform similarity (WSOLA) for high quality time-scale modification of speech , 1993, 1993 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[232]  B C Moore,et al.  Spectral contrast enhancement of speech in noise for listeners with sensorineural hearing impairment: effects on intelligibility, quality, and response times. , 1993, Journal of rehabilitation research and development.

[233]  O Dyrlund,et al.  Acoustic feedback margin improvements in hearing instruments using a prototype DFS (digital feedback suppression) system. , 1991, Scandinavian audiology.

[234]  Harry L. Van Trees,et al.  Detection, Estimation, and Modulation Theory, Part I , 1968 .

[235]  W Soede,et al.  Development of a directional hearing instrument based on array technology. , 1993, The Journal of the Acoustical Society of America.

[236]  E. Shaw The External Ear , 1974 .

[237]  Roger Lagadec,et al.  Signal Enhancement via Digital Signal Processing , 1983 .

[238]  C. Chui Wavelet Analysis and Its Applications , 1992 .

[239]  Xavier Rodet,et al.  Analysis of Sound for Additive Synthesis: Tracking of Partials Using Hidden Markov Models , 1993, ICMC.

[240]  Arlene C. Neuman,et al.  Noise reduction for hearing aids , 1989, Proceedings of the Fifteenth Annual Northeast Bioengineering Conference.

[241]  D T Lawson,et al.  Design and evaluation of a continuous interleaved sampling (CIS) processing strategy for multichannel cochlear implants. , 1993, Journal of rehabilitation research and development.

[242]  G. Fairbanks,et al.  Method for time of frequency compression-expansion of speech , 1954 .

[243]  M P Haggard,et al.  Two-state compression of spectral tilt: individual differences and psychoacoustical limitations to the benefit from compression. , 1987, Journal of rehabilitation research and development.

[244]  Julius O. Smith,et al.  The one-filter Keefe clarinet tonehole , 1997, Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics.

[245]  Thomas Sporer,et al.  -NMR- and -Masking Flag-: Evaluation of Quality Using Perceptual Criteria , 1992 .

[246]  R.W. Schafer,et al.  Constrained iterative restoration algorithms , 1981, Proceedings of the IEEE.

[247]  P. Woodland,et al.  A computational model of the auditory periphery for speech and hearing research. II. Descending paths. , 1994, The Journal of the Acoustical Society of America.

[248]  Bernd Edler,et al.  LINC: a common theory of transform and subband coding , 1993, IEEE Trans. Commun..

[249]  Julius O. Smith A new approach to digital reverberation using closed waveguide networks , 1985 .

[250]  Frank Baumgarte,et al.  A Nonlinear Psychoacoustic Model Applied to ISO/MPEG Layer 3 Coder , 1995 .

[251]  Neil L. Gerr,et al.  The Generalised Spectrum and Spectral Coherence of a Harmonizable Time Series , 1994 .

[252]  I. I. Eldumiati,et al.  Digital signal processor: Architecture and performance , 1981, The Bell System Technical Journal.

[253]  B. Atal,et al.  Optimizing digital speech coders by exploiting masking properties of the human ear , 1978 .

[254]  W. L. Miranker,et al.  The recovery of distorted band-limited signals , 1961 .

[255]  D. B. Preston Spectral Analysis and Time Series , 1983 .

[256]  H. Hake,et al.  On the Masking Pattern of a Simple Auditory Stimulus , 1950 .

[257]  Amir Dembo,et al.  Signal synthesis from modified discrete short-time transform , 1988, IEEE Trans. Acoust. Speech Signal Process..

[258]  James A. Moorer,et al.  About This Reverberation Business , 1978 .

[259]  Stephen C. Glinski,et al.  Spoken Language Recognition on a DSP Array Processor , 1994, IEEE Trans. Parallel Distributed Syst..

[260]  John G. Beerends,et al.  A Perceptual Audio Quality Measure Based on a Psychoacoustic Sound Representation , 1992 .

[261]  L. J. Griffiths,et al.  An alternative approach to linearly constrained adaptive beamforming , 1982 .

[262]  Julius O. Smith,et al.  Physical Modeling with the 2-D Digital Waveguide Mesh , 1993, ICMC.

[263]  Julius O. Smith,et al.  Physical Modeling Synthesis Update , 1996 .

[264]  Miller Puckette,et al.  The Architecture of the IRCAM Musical Workstation , 1991, USENIX Summer.

[265]  Jean-Marc Jot,et al.  Digital Delay Networks for Designing Artificial Reverberators , 1991 .

[266]  E. Zwicker,et al.  Audio engineering and psychoacoustics: matching signals to the final receiver, the human auditory system , 1991 .

[267]  Francis F. Lee,et al.  Time Compression and Expansion of Speech by the Sampling Method , 1972 .

[268]  R H Brey,et al.  Application of adaptive digital signal processing to speech enhancement for the hearing impaired. , 1987, Journal of rehabilitation research and development.

[269]  O. Cappé,et al.  Regularized estimation of cepstrum envelope from discrete frequency points , 1995, Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics.

[270]  R Plomp,et al.  The negative effect of amplitude compression in multichannel hearing aids in the light of the modulation-transfer function. , 1988, The Journal of the Acoustical Society of America.

[271]  Unto K. Laine,et al.  Splitting the Unit Delay - Tools for fractional delay filter design , 1996 .

[272]  Hugo Fastl,et al.  Psychoacoustics: Facts and Models , 1990 .

[273]  Günther Theile,et al.  Low-Bit Rate Coding of High Quality Audio Signals , 1987 .

[274]  T. Quatieri,et al.  Phase modelling and its application to sinusoidal transform coding , 1986, ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[275]  Peter J. W. Rayner,et al.  Separation of stationary and time-varying systems and its application to the restoration of gramophone recordings , 1989, IEEE International Symposium on Circuits and Systems,.

[276]  John Vanderkooy,et al.  Digital Dither: Processing with Resolution Far Below the Least Significant Bit , 1989 .

[277]  J M Aran,et al.  AP tuning curves from normal and pathological human and guinea pig cochleas. , 1981, The Journal of the Acoustical Society of America.

[278]  Xavier Serra,et al.  A sound analysis/synthesis system based on a deterministic plus stochastic decomposition , 1990 .

[279]  P. E. Axon,et al.  A study of frequency fluctuations in sound recording and reproducing systems , 1949 .

[280]  Simon J. Godsill,et al.  Bayesian model selection for time series using Markov chain Monte Carlo , 1997, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[281]  R C Seewald,et al.  Ear level recordings of the long-term average spectrum of speech. , 1991, Ear and hearing.

[282]  Richard Jamss Pinnell Adaptive Transform Coding of Speech Signals , 1982 .

[283]  W. M. Rabinowitz,et al.  On the potential of fixed arrays for hearing aids. , 1993, The Journal of the Acoustical Society of America.

[284]  R.F. Kubichek,et al.  Speech quality assessment using expert pattern recognition , 1989, Conference Proceeding IEEE Pacific Rim Conference on Communications, Computers and Signal Processing.

[285]  John Princen,et al.  Subband/Transform coding using filter bank designs based on time domain aliasing cancellation , 1987, ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[286]  A. W. M. van den Enden,et al.  Discrete Time Signal Processing , 1989 .

[287]  Daniel Arfib,et al.  Musical Transformations Using the Modification of Time-Frequency Images , 1993 .

[288]  H J McDermott,et al.  A new portable sound processor for the University of Melbourne/Nucleus Limited multielectrode cochlear implant. , 1992, The Journal of the Acoustical Society of America.

[289]  M. Schroeder Number Theory in Science and Communication , 1984 .

[290]  A. Gray,et al.  A normalized digital filter structure , 1975 .

[291]  Thomas F. Quatieri,et al.  Phase coherence in speech reconstruction for enhancement and coding applications , 1989, International Conference on Acoustics, Speech, and Signal Processing,.

[292]  B C Moore,et al.  Spectral feature enhancement for people with sensorineural hearing impairment: effects on speech intelligibility and quality. , 1992, Journal of rehabilitation research and development.

[293]  So,et al.  An excitation‐pattern model for intensity discrimination , 1981 .

[294]  James David Johnston,et al.  Enhancing the Performance of Perceptual Audio Coders by Using Temporal Noise Shaping (TNS) , 1996 .

[295]  M. Schroeder,et al.  On Frequency Response Curves in Rooms. Comparison of Experimental, Theoretical, and Monte Carlo Results for the Average Frequency Spacing between Maxima , 1962 .

[296]  Mark R. Weiss,et al.  Automatic Detection and Enhancement of Speech Signals , 1975 .

[297]  John H. L. Hansen,et al.  Discrete-Time Processing of Speech Signals , 1993 .

[298]  Raymond N. J. Veldhuis,et al.  Subband coding of stereophonic digital audio signals , 1991, [Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing.

[299]  Adrian F. M. Smith,et al.  Sampling-Based Approaches to Calculating Marginal Densities , 1990 .

[300]  M. Sachs,et al.  Two-tone inhibition in auditory-nerve fibers. , 1968, The Journal of the Acoustical Society of America.

[301]  Jerry D. Gibson,et al.  Digital coding of waveforms: Principles and applications to speech and video , 1985, Proceedings of the IEEE.

[302]  Ray Meddis,et al.  Virtual pitch and phase sensitivity of a computer model of the auditory periphery , 1991 .

[303]  Jean Laroche,et al.  Evaluation of short-time spectral attenuation techniques for the restoration of musical recordings , 1995, IEEE Trans. Speech Audio Process..

[304]  Julius O. Smith,et al.  Digital Waveguide Modeling of Woodwind Toneholes , 1997, ICMC.

[305]  Louis Dunn Fielder,et al.  AC-2 and AC-3: Low-Complexity Transform-Based Audio Coding , 1996 .

[306]  D B Hawkins,et al.  Signal-to-noise ratio advantage of binaural hearing aids and directional microphones under different levels of reverberation. , 1984, The Journal of speech and hearing disorders.

[307]  P. Fitzgibbons,et al.  Gap detection in normal and hearing-impaired listeners. , 1982, The Journal of the Acoustical Society of America.

[308]  S. Mitra,et al.  A unified approach to time- and frequency-domain realization of FIR adaptive digital filters , 1983 .

[309]  Jon Emil Natvig Evaluation of six medium bit-rate coders for the Pan-European digital mobile radio system , 1988, IEEE J. Sel. Areas Commun..

[310]  A. El-Jaroudi,et al.  Time-scale modification in medium to low rate speech coding , 1986, ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[311]  J. Moorer The Synthesis of Complex Audio Spectra by Means of Discrete Summation Formulas , 1976 .

[312]  Stephen McAdams,et al.  Spectral fusion, spectral parsing and the formation of auditory images , 1984 .

[313]  M R Leek,et al.  Learning to detect auditory pattern components. , 1984, The Journal of the Acoustical Society of America.

[314]  G. G. Stokes "J." , 1890, The New Yale Book of Quotations.

[315]  Hideo Suzuki,et al.  Electronic musical instrument forming tones by wave computation , 1987 .

[316]  R. Gray,et al.  Distortion measures for speech processing , 1980 .

[317]  Peter R. Samson,et al.  A General-Purpose Digital Synthesizer , 1980 .

[318]  Thomas G. Stockham,et al.  A-D and D-A Converters: Their Effect on Digital Audio Fidelity , 1971 .

[319]  A. Krokstad,et al.  Calculating the acoustical room response by the use of a ray tracing technique , 1968 .

[320]  Schuyler Quackenbush,et al.  Objective measures of speech quality , 1995 .

[321]  이종화,et al.  오디오 신호 처리용 디지털 시그날 프로세서의 VLSI 설계 ( VLSI Design of a Digital Signal Processor for Audio Applications ) , 1995 .

[322]  Takis Kasparis,et al.  Adaptive scratch noise filtering , 1993 .

[323]  D. K. Bustamante,et al.  Measurement and adaptive suppression of acoustic feedback in hearing aids , 1989, International Conference on Acoustics, Speech, and Signal Processing,.

[324]  D. Malah,et al.  Frequency scaling of speech signals by transform techniques , 1981, The Bell System Technical Journal.

[325]  Akihiko Sugiyama,et al.  An efficient tonal component coding algorithm for MPEG-2 Audio NBC , 1997, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[326]  L. B. Jackson,et al.  An approach to the implementation of digital filters , 1968 .

[327]  R. Bruce Lindsay,et al.  Acoustics: historical and philosophical development , 1973 .

[328]  I. J. Leontaritis,et al.  Model selection and validation methods for non-linear systems , 1987 .

[329]  Simon J. Godsill,et al.  The restoration of pitch variation defects in gramophone recordings , 1993, Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics.

[330]  T. M. Cannon,et al.  Blind deconvolution through digital signal processing , 1975, Proceedings of the IEEE.

[331]  T. W. Parks,et al.  Digital Filter Design , 1987 .

[332]  Ajm Adrian Houtsma,et al.  Loudness, pitch, localization, aural distortion, pathology , 1986 .

[333]  Panos Papamichalis,et al.  The TMS320C30 floating-point digital signal processor , 1988, IEEE Micro.

[334]  Wj Fitzgerald,et al.  The restoration of digital audio recordings using the Gibbs sampler , 1993 .

[335]  Tapio Takala,et al.  Waveguide Mesh Method for Low-Frequency Simulation of Room Acoustics , 1995 .

[336]  A A Montgomery,et al.  Evaluation of two speech enhancement techniques to improve intelligibility for hearing-impaired adults. , 1988, Journal of speech and hearing research.

[337]  Maciej Niedzwiecki,et al.  Adaptive scheme for elimination of broadband noise and impulsive disturbances from AR and ARMA signals , 1996, IEEE Trans. Signal Process..

[338]  Carmen García-Mateo,et al.  Shape-invariant pitch-synchronous text-to-speech conversion , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.

[339]  B. Paillard,et al.  PERCEVAL: Perceptual Evaluation of the Quality of Audio Signals , 1992 .

[340]  Norbert Wiener,et al.  Extrapolation, Interpolation, and Smoothing of Stationary Time Series, with Engineering Applications , 1949 .

[341]  Takehiro Moriya,et al.  High-quality audio coding at less than 64 kbit/s by using TwinVQ , 1995 .

[342]  Thomas F. Quatieri,et al.  Shape invariant time-scale and pitch modification of speech , 1992, IEEE Trans. Signal Process..

[343]  R. Simar,et al.  The TMS320 family of digital signal processors , 1987, Proceedings of the IEEE.

[344]  C M Reed,et al.  Hearing aids--a review of past research on linear amplification, amplitude compression, and frequency lowering. , 1979, ASHA monographs.

[345]  J. Flanagan,et al.  Synthesis of voiced sounds from a two-mass model of the vocal cords , 1972 .

[346]  Matti Karjalainen,et al.  Digital Waveguide Modeling of Wind Instrument Bores constructed of Truncated Cones , 1994, ICMC.

[347]  Sheng Chen,et al.  Modelling and analysis of non-linear time series , 1989 .

[348]  E.A. Lee Programmable DSP architectures. II , 1989, IEEE ASSP Magazine.

[349]  Sailes K. Sengijpta Fundamentals of Statistical Signal Processing: Estimation Theory , 1995 .

[350]  S. Thomas Alexander,et al.  Adaptive Signal Processing , 1986, Texts and Monographs in Computer Science.

[351]  William G. Gardner,et al.  Efficient Convolution without Input/Output Delay , 1995 .

[352]  Davide Rocchesso,et al.  Circulant Feedback Delay Networks for Sound Synthesis and Processing , 1994, ICMC.

[353]  Thomas F. Quatieri,et al.  Peak-to-RMS reduction of speech based on a sinusoidal model , 1991, IEEE Trans. Signal Process..

[354]  Douglas Preis,et al.  Perception of Phase Distortion in Anti-Alias Filters , 1984 .

[355]  P. Vaidyanathan Multirate Systems And Filter Banks , 1992 .

[356]  R A Lutfi A power-law transformation predicting masking by sounds with complex spectra. , 1985, The Journal of the Acoustical Society of America.

[357]  John G. Beerends,et al.  The Role of International Masking and Perceptual Streaming in the Measurement of Music Codec Quality , 1996 .

[358]  P. Yip,et al.  Discrete Cosine Transform: Algorithms, Advantages, Applications , 1990 .

[359]  John Vanderkooy,et al.  Correction to "Resolution Below the Least Significant Bit in Digital Systems with Dither" , 1984 .

[360]  J. P. Norton,et al.  An Introduction to Identification , 1986 .

[361]  Frederick Mosteller,et al.  Data Analysis and Regression , 1978 .

[362]  J M Festen,et al.  The effect of frequency-selective attenuation on the speech-reception threshold of sentences in conditions of low-frequency noise. , 1991, The Journal of the Acoustical Society of America.

[363]  B. Widrow,et al.  Adaptive noise cancelling: Principles and applications , 1975 .

[364]  Jean-Marc Jot,et al.  Digital Signal Processing Issues in the Context of Binaural and Transaural Stereophony , 1995 .

[365]  A. Chaigne,et al.  Numerical simulations of piano strings. I. A physical model for a struck string using finite difference methods , 1994 .

[366]  Thomas W. Parsons,et al.  Study and Development of the INTEL Technique for Improving Speech Intelligibility , 1975 .

[367]  S. Schwerman,et al.  The Physics of Musical Instruments , 1991 .

[368]  Jae S. Lim,et al.  Multiband excitation vocoder , 1988, IEEE Transactions on Acoustics, Speech, and Signal Processing.

[369]  Ronald E. Crochiere,et al.  Real-Time Implementation of Time Domain Harmonic Scaling of Speech for Rate Modification and Coding , 1983 .

[370]  M Kompis,et al.  Noise reduction for hearing aids: combining directional microphones with an adaptive beamformer. , 1994, The Journal of the Acoustical Society of America.

[371]  S Buus,et al.  Temporal gap detection in sensorineural and simulated hearing impairments. , 1984, Journal of speech and hearing research.

[372]  S. Boll,et al.  Techniques for suppression of an interfering talker in co-channel speech , 1987, ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[373]  Man Mohan Sondhi,et al.  A hybrid time-frequency domain articulatory speech synthesizer , 1987, IEEE Trans. Acoust. Speech Signal Process..

[374]  Julius O. Smith,et al.  Efficient Simulation of the Reed-Bore and Bow-String Mechanisms , 1986, ICMC.

[375]  J. Makhoul,et al.  Linear prediction: A tutorial review , 1975, Proceedings of the IEEE.

[376]  R. Douglas Martin,et al.  ROBUST METHODS FOR TIME SERIES , 1981 .

[377]  Simon J. Godsill,et al.  Frequency-domain interpolation of sampled signals , 1993 .

[378]  Robert B. Dunn,et al.  Signal Enhancement in AM-FM Interference , 1994 .

[379]  F. Richard Moore,et al.  Elements of computer music , 1990 .

[380]  A. Wilgus,et al.  High quality time-scale modification for speech , 1985, ICASSP '85. IEEE International Conference on Acoustics, Speech, and Signal Processing.

[381]  Julius O. Smith,et al.  A flexible sampling-rate conversion method , 1984, ICASSP.

[382]  John Mick,et al.  Bit-slice Microprocessor Design , 1980 .

[383]  E W Yund,et al.  Speech discrimination with an 8-channel compression hearing aid and conventional aids in background of speech-band noise. , 1987, Journal of rehabilitation research and development.

[384]  Simon J. Godsill,et al.  Robust noise modelling with application to audio restoration , 1995, Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics.

[385]  Louis Dunn Fielder,et al.  ISO/IEC MPEG-2 Advanced Audio Coding , 1997 .

[386]  G. K. Yates,et al.  Nonlinear input-output functions derived from the responses of guinea-pig cochlear nerve fibres: Variations with characteristic frequency , 1994, Hearing Research.

[387]  Thomas Kailath,et al.  On spatial smoothing for direction-of-arrival estimation of coherent signals , 1985, IEEE Trans. Acoust. Speech Signal Process..

[388]  B. Gold,et al.  A digital frequency synthesizer , 1971 .

[389]  Giorgio Dimino,et al.  Entropy Reduction in High-Quality Audio Coding , 1995 .

[390]  Luís B. Almeida,et al.  Variable-frequency synthesis: An improved harmonic coding scheme , 1984, ICASSP.

[391]  Gerald Schuller A Low-Delay Filter Bank for Audio Coding with Reduced Pre-Echoes , 1995 .

[392]  Jae S. Lim,et al.  The unimportance of phase in speech enhancement , 1982 .

[393]  Robert B. Dunn,et al.  Time-scale modification with temporal envelope invariance , 1993, Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics.

[394]  J. L. Flanagan,et al.  Automatic generation of voiceless excitation in a vocal-cord/vocal-tract speech synthesizer , 1975 .

[395]  G.D. Cain,et al.  Approximation of FIR by IIR digital filters: an algorithm based on balanced model reduction , 1992, IEEE Trans. Signal Process..

[396]  Manfred R. Schroeder,et al.  Bandwidth compression of speech by analytic-signal rooting , 1967 .

[397]  D. V. Maercke,et al.  Binaural simulation of concert halls: A new approach for the binaural reverberation process , 1993 .

[398]  John W. Tukey,et al.  Exploratory Data Analysis. , 1979 .

[399]  Bryan Holloway,et al.  Timbre morphing of sounds with unequal numbers of features , 1995 .

[400]  Eric Moulines,et al.  HNS: Speech modification based on a harmonic+noise model , 1993, 1993 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[401]  Man Mohan Sondhi,et al.  Techniques for estimating vocal-tract shapes from the speech signal , 1994, IEEE Trans. Speech Audio Process..

[402]  D. Paul The spectral envelope estimation vocoder , 1981 .

[403]  L. Braida,et al.  Towards a model for discrimination of broadband signals. , 1986, The Journal of the Acoustical Society of America.

[404]  Simon J. Godsill,et al.  Robust reconstruction and analysis of autoregressive signals in impulsive noise signals using Gibbs sampler , 1995 .

[405]  Søren Holdt Jensen,et al.  Reduction of broad-band noise in speech by truncated QSVD , 1995, IEEE Trans. Speech Audio Process..

[406]  George S. Moschytz,et al.  Noise reduction by noise-adaptive spectral magnitude expansion , 1994 .

[407]  David Malah,et al.  Speech enhancement using a minimum mean-square error log-spectral amplitude estimator , 1984, IEEE Trans. Acoust. Speech Signal Process..

[408]  Andrew Sekey,et al.  An Objective Measure for Predicting Subjective Quality of Speech Coders , 1992, IEEE J. Sel. Areas Commun..

[409]  Yrjö Neuvo,et al.  Novel floating-point A/D- and D/A-conversion methods , 1994, Proceedings of IEEE International Symposium on Circuits and Systems - ISCAS '94.

[410]  M. Liberman,et al.  Single-neuron labeling and chronic cochlear pathology. III. Stereocilia damage and alterations of threshold tuning curves , 1984, Hearing Research.

[411]  Louis Dunn Fielder The audibility of modulation noise in floating-point conversion systems , 1985 .

[412]  P. Depalle,et al.  Spectral Envelopes and Inverse FFT Synthesis , 1992 .

[413]  Miller S. Puckette,et al.  Designing Multi-Channel Reverberators , 1982 .

[414]  Yuan‐Hwang Chen,et al.  Frequency‐domain implementation of Griffiths–Jim adaptive beamformer , 1992 .

[415]  Mark Kahrs,et al.  Gnot Music: A Flexible Workstation for Orchestral Synthesis , 1992 .

[416]  Y. Mahieux,et al.  Transform coding of audio signals at 64 kbit/s , 1990, [Proceedings] GLOBECOM '90: IEEE Global Telecommunications Conference and Exhibition.

[417]  Franklin Richard Moore,et al.  Real time interactive computer music synthesis , 1977 .

[418]  A. Gray,et al.  Distance measures for speech processing , 1976 .

[419]  John M. Snell Multiprocessor DSP Architectures and Implications for Software , 1989 .

[420]  Max V. Mathews,et al.  GROOVE—a program to compose, store, and edit functions of time , 1970, CACM.

[421]  R. T. Schumacher,et al.  ON THE OSCILLATIONS OF MUSICAL-INSTRUMENTS , 1983 .

[422]  Stephen A. Dyer,et al.  Digital signal processing , 2018, 8th International Multitopic Conference, 2004. Proceedings of INMIC 2004..

[423]  Mark A. Clements,et al.  Speech concatenation and synthesis using an overlap-add sinusoidal model , 1996, 1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings.

[424]  B Kollmeier,et al.  Real-time multiband dynamic compression and noise reduction for binaural hearing aids. , 1993, Journal of rehabilitation research and development.

[425]  Gérard Faucon,et al.  A Perceptual Objective Measurement System (POM) for the Quality Assessment of Perceptual Codecs , 1994 .

[426]  Donald Byrd,et al.  The Kurzweil 250 Digital Synthesizer , 1986 .

[427]  Alan V. Oppenheim,et al.  All-pole modeling of degraded speech , 1978 .

[428]  David P. Berners On the Use of Schrodinger's Equation in the Analytic Determination of Horn Reflectance , 1994, ICMC.

[429]  Ernst Terhardt,et al.  Calculating virtual pitch , 1979, Hearing Research.

[430]  Thomas F. Quatieri,et al.  Speech analysis/Synthesis based on a sinusoidal representation , 1986, IEEE Trans. Acoust. Speech Signal Process..

[431]  Peter Kabal,et al.  Time-scale modification of speech using an incremental time-frequency approach with waveform structure compensation , 1992, [Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing.

[432]  Xavier Serra,et al.  A system for sound analysis/transformation/synthesis based on a deterministic plus stochastic decomposition , 1989 .

[433]  G. C. Tiao,et al.  A bayesian approach to some outlier problems. , 1968, Biometrika.

[434]  A. Oppenheim,et al.  Signal reconstruction from phase or magnitude , 1980 .

[435]  Kenneth J. Pope,et al.  Non-linear system identification using Bayesian inference , 1994, Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing.

[436]  B Hagerman,et al.  Questionnaires on desirable properties of hearing aids. , 1985, Scandinavian audiology.

[437]  Eric Moulines,et al.  Pitch-synchronous waveform processing techniques for text-to-speech synthesis using diphones , 1989, Speech Commun..

[438]  P M Zurek,et al.  Evaluation of an adaptive beamforming method for hearing aids. , 1992, The Journal of the Acoustical Society of America.

[439]  J. B. Millar,et al.  A preliminary report on a multiple-channel cochlear implant operation , 1979, The Journal of Laryngology & Otology.

[440]  Pjw Rayner,et al.  A new application of adaptive filters for restoration of archived gramophone recordings , 1988, ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing.

[441]  J. Beauchamp,et al.  An investigation of vocal vibrato for synthesis , 1990 .

[442]  Yannis Stylianou,et al.  HNM: a simple, efficient harmonic+noise model for speech , 1993, Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics.

[443]  Jae Lim,et al.  Signal reconstruction from short-time Fourier transform magnitude , 1983 .