Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics最新文献

{"title":"Chaos In Aperiodicity Of Musical Oscillators","authors":"C. Chafe","doi":"10.1109/ASPAA.1991.634152","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634152","url":null,"abstract":"The aperiodicity characteristic of many self-sustained musical instruments like bowed strings, voice, woodwinds or brass, reveals certain chaotic structures when observed over many periods. Short-lived subharmonics are often detectable and these are thought to be the result of at least four general properties of the iinstruments: complex resonance paths, limit-cycles and phase transition boundaries in the feeback mechanism and pulsed noise in the excitation mechxnism. Examples from real data and simulations isolating these phenomena in physical models simulations will be compared. The conclusions point to principles that can be applied to music synthesis methods. Phase portraits of recorded instrument tones can be animated in time to display the characteristics of aperidocity in a meaningful way. It is seen that certain portions of the waveform are more variable from period-toperiod than other portions. Through time, the variation exhibits a degree of repetitive structure that gives rise to perceptible noisy subharmonics. One method for portraying subharmonic activity is to display succesive periods as raster lines in an oblong plot of phase vs. period. Gray-level is used to display the variations observed in phase portraits. The best sensitivity to this variation has been acheived by plotting period-to-period vector length differences where the vector is the distance between two samples in the phase portrait. Subharmonics arise from several possible mechanisms. Trombone tones have been analyzed with the method and show a correlation between overblown harmonic number and subharmonic number. For example, a fourth harmonic shows distinct fourth subharmonics in its raster plot. The explanation is that the fundamental round-trip still contributes to the system even","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115642369","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"On Floaking-Point Sigma-Delta D/A Conversion","authors":"J. Kontro, B. Zeng, Y. Neuvo","doi":"10.1109/ASPAA.1991.634128","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634128","url":null,"abstract":"Digital audio has rapidly replaced andog audio ining the demand for high-quality analog-tcwdigital (A/D) and digital-to-analog (D/A) converters. The pmformance of conventional Nyquist rate PCM converters is limited by the need of a high-order analog anti-aliasing filter, a reconstruction filter and a sample-and-hold amplifier. These limitations CM be eliminated and better performance can be achieved by using oversampled sigmadelta (SA) converters which convert the signal to a high-frequency one-bit streem. S A converters utilize a noise shaping feature in which quantization errors are shaped to high fiequencies and removed with a digital lowpass filter [1]-131. Fkently, oversampled CA converters have emerged in numerous digital audio equipments, such as CD and DAT players. Currently high quality audio converters are based on uniform, fixed-point quantization scheme, its the dynamic range and the signal-to-noise ratio (SNR) depend on the conversion accuracy. The SNX depends also on the signal level; hence it decreases with low signal levels. Unifonn quantization is suitable for signals which distribute evenly in the converter amplitude range. Music signals do not however have a uniform distribution and therefore new quantization schemes are desired if a better SNR behavior is wanted. Better performance can be obtained if the quantization level are spread approximately logarithmically. This can be irchieved by using compending converters or floating-point converters. Examples of non-uniform quantizers are found in PCM telephones, and NICAM [4]-[5] and DIGICIPHERTM [SI television sound systems. In recent years, the use of floating-point arithmetic in digital signal processing (DSP) has rapidly i n c r e a d due to the development of fast and low-cost floating-point signal processors. Since the algorithms are based on floating-point arithmetic, a need for floating-point converters has arisen. Eventually even such functions as volume control are likely to be implemented in digital form requiring a larger dynamic range for the D/A converter than is today commonly used. Even there are some difficulties in implementing Nyquist rate floating-point converters, they have beein studied extensively [A, [SI. In this paper, we discuss the design and implementation aspects of oversempled C A D/A conversion based on a full floating-point number system.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114166652","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"A perceptual representation of audio for co-channel source separation","authors":"D. Ellis, B. Vercoe, T. Quatieri","doi":"10.1109/ASPAA.1991.634109","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634109","url":null,"abstract":"Jntroduction Despite the many advances in signal processing mathematics in recent years, the greatest gains and breakthroughs are made by exploiting special properties of the particular problem at hand. Since very often the ultimate destination of processed sound is a human listener, the many complex interactions and constraints of the auditory system are available for exploitation. However, these constraints are so involved that we are only just beginning to understand the possibilities they offer.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116544181","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"Sound Synthesis Of Stringed Instruments Using Statistical Modeling Of The Input Admittance","authors":"A. Chaigne","doi":"10.1109/ASPAA.1991.634149","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634149","url":null,"abstract":"For stringed instruments such as guitar, cello or violin, the term \"input admittance\" (IA) refers to the driving point mobility. This quantity is obtained by simultaneous measurements of force and velocity (or weleration) at a carefully selected point, near the bridge [l]. Such measurements have been used for many years for characterizing the quality of the instruments. However, the question whether the measured data are significant from an audible point of view remains still today a subject of controversy. Therefore it is of great interest to include the IA in a synthesis p r o p m based on physical modeling, so as to validate its perceptual relevance. The main features of a typical accelerance (acceleration/driving force vs. frequency) modulus cunie can be clearly Seen in Fig. 1. This curve exhibits well separated peaks in the low-frequency range, whereas the high-frequency range is more continuous. In this later region the bandwidths of the different resonance:; overlap, and one must use modal density and statistical parameters rather than individual modal quantities in ordeir to describe the vibration properties of the body.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122310591","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"Auditory Representation and Sound Separation","authors":"M. Slaney, R. Lyon","doi":"10.1109/ASPAA.1991.634089","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634089","url":null,"abstract":"The brain uses several clues for sound separation. Binaural Ilocalization, onsets and common modulation are some of the more important ones. To this end, we have been exploring the use of a perceptually-motivated threedimensional representation of sound called the correlogram. The correlogram represents sound as a moving image of cochlear place (or frequency) and short-time autocorrelation versus time. The result i3 a compelling visualization of sound, which encodes many of the perceptually important clues in a form where these clues are easy to detect.","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128179769","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}

{"title":"Generalized Discrete Cepstral Analysis for Decorrvolution of Source-Filter System with Discrete Spectra","authors":"T. Galas, X. Rodet","doi":"10.1109/ASPAA.1991.634108","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634108","url":null,"abstract":"","PeriodicalId":146017,"journal":{"name":"Final Program and Paper Summaries 1991 IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"1900-01-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"113939639","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}