Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634152
C. Chafe
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
{"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}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634128
J. Kontro, B. Zeng, Y. Neuvo
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.
由于对高质量模数转换器(A/D)和数模转换器(D/A)的需求，数字音频已经迅速取代了普通音频。传统奈奎斯特速率PCM变换器的性能受到高阶模拟抗混叠滤波器、重构滤波器和采样保持放大器的限制。通过使用过采样信号(SA)转换器将信号转换为高频1位流，可以消除这些限制并获得更好的性能。S A转换器利用噪声整形特性，将量化误差塑造为高频，并通过数字低通滤波器去除[1]-131。近年来，过采样CA转换器已经出现在许多数字音频设备中，如CD和DAT播放器。目前高质量的音频转换器都是基于均匀的定点量化方案，其动态范围和信噪比取决于转换精度。SNX也取决于信号电平;因此它随着低信号电平而减小。均匀量化适用于在变换器幅度范围内均匀分布的信号。然而，音乐信号不具有均匀分布，因此，如果需要更好的信噪比行为，则需要新的量化方案。如果量化水平近似对数扩展，则可以获得更好的性能。这可以通过使用补偿转换器或浮点转换器来实现。非均匀量化器的例子见于PCM电话、NICAM[4]-[5]和DIGICIPHERTM [SI电视音响系统]。近年来，由于快速、低成本的浮点信号处理器的发展，浮点运算在数字信号处理(DSP)中的应用得到了迅速的发展。由于这些算法是基于浮点运算的，因此出现了对浮点转换器的需求。最终，甚至像音量控制这样的功能也可能以数字形式实现，需要比目前常用的D/ a转换器更大的动态范围。尽管在实现奈奎斯特率浮点转换器时存在一些困难，但它们已被广泛研究[A， [SI]。本文讨论了基于全浮点数系统的过采样C - A - D/A转换的设计与实现。
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Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634109
D. Ellis, B. Vercoe, T. Quatieri
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.
{"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}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634149
A. Chaigne
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.
{"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}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634089
M. Slaney, R. Lyon
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.
{"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}
Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634108
T. Galas, X. Rodet
{"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}
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