Pub Date : 1900-01-01DOI: 10.1109/ASPAA.1991.634097
M. F. Davis
Contemporary analog and digital signal processing techniques have 'been refined to the point of being able to routinely convey individual audio channels with little or no perceptible loss of quality. The greatest disparity between original anti reproduced soundfields usually involves their spatial characteristics. Improving the spatial fidelity of audio recording and transmission systems involves understanding the underlying localization mechanisms, then applying this understanding to evolve specific system requirements, subject to the constraints of real world components and practices. Spatial audio systems are conveniently dividable into three functional blocks: 1. soundfield pickup, via one or more microphones and/or electronically synthesized signals, 2. means for coding, transmission (or recording/playback), and decoding of the net source audio signals, and 3. soundfield reconstruction, via loudspeakers or headphones, and possible associated processing. The presentation environment exerts sufficient influence on system configuration to have spawned several classes of spatial audio systems, e.g. home, headphone, and theatre systems. Conventional home stereo is currently the most common spatial audio system in use. It purports to encode a horizontal continuum of space into a pair of audio channels, which are then conveyed via a discrete two channel medium to a pair of loudspeakers. This system relies on the psychoacoustic phenomenon of phantom images; to try to fill the space between the speakers. Related techniques, such as 'Sonic Holography' or Q-Sound, attempt to extend the range of horizontal space conveyed, in part by using interaural cross cancellation to extend the apparent reproduced image beyond the arc of the speakers. Microphone pickup arrangements for stereo recording vary widely, and are often a matter of strong individual preference on the part of recording producers. It is desirable for newly developed systems to retain this option.
{"title":"Spatial Sound Recording and Transmission Systems: Status, Problems, and Prospects","authors":"M. F. Davis","doi":"10.1109/ASPAA.1991.634097","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634097","url":null,"abstract":"Contemporary analog and digital signal processing techniques have 'been refined to the point of being able to routinely convey individual audio channels with little or no perceptible loss of quality. The greatest disparity between original anti reproduced soundfields usually involves their spatial characteristics. Improving the spatial fidelity of audio recording and transmission systems involves understanding the underlying localization mechanisms, then applying this understanding to evolve specific system requirements, subject to the constraints of real world components and practices. Spatial audio systems are conveniently dividable into three functional blocks: 1. soundfield pickup, via one or more microphones and/or electronically synthesized signals, 2. means for coding, transmission (or recording/playback), and decoding of the net source audio signals, and 3. soundfield reconstruction, via loudspeakers or headphones, and possible associated processing. The presentation environment exerts sufficient influence on system configuration to have spawned several classes of spatial audio systems, e.g. home, headphone, and theatre systems. Conventional home stereo is currently the most common spatial audio system in use. It purports to encode a horizontal continuum of space into a pair of audio channels, which are then conveyed via a discrete two channel medium to a pair of loudspeakers. This system relies on the psychoacoustic phenomenon of phantom images; to try to fill the space between the speakers. Related techniques, such as 'Sonic Holography' or Q-Sound, attempt to extend the range of horizontal space conveyed, in part by using interaural cross cancellation to extend the apparent reproduced image beyond the arc of the speakers. Microphone pickup arrangements for stereo recording vary widely, and are often a matter of strong individual preference on the part of recording producers. It is desirable for newly developed systems to retain this option.","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":"126839359","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.1991.634102
Theodore J Doll, Thomas E Hanna
It has been suggested that three dimensional (3-D) auditory displays could enhance operator performance in a wide variety of applications, including sonar (Doll, Hanna, and Ruissotti, in press), auditory warnings in aircraft cockpits (Doll et al., Most of the anticipated applications of 3-D auditory dislplays involve simultaneous presentation of multiple signals from different directions. A potential problem is that signals that are sounded simultaneously or closely in time may mask one-another. The extent to which simultaneous sounds mask one-(another should depend both upon their spectral similarity and how closely their sources are positioned in space. It is well esta'blished that masking is greatly reduced when the masker and signal do not occupy the same critical band (e.g., Durlach & Colburn, 1978). Studies of free-field masking show that the effectiveness of a masker decreases as it is separated in space from the signal However, the extent to which spectral and spatial similarity trade-off in determining the detectability of signals i n 3-0 auditory displays is unknom. This information i.s needed to design effective 3-D displays. The purpose of this research was to deternine how the spectral and spatial similarity of signals arid naskers interact to determine the deteczability of s i p a i s in 3-D audicory displays. A tonal signal and a **notchedtg noise masker were presented from loudspeakers with various spatial separations (0, 20, and 40 degrees) in a free field (i.e. , a *treal*t 3-D auditory display). The loudspeakers were arranged in a horizontal circular arc 10 ft. from the listener at ear l e v e l. The spectral similarity of the masker and signal were manipulated by varying the low-pass cutoff of one noise band and the high-pass cutoff of another, independent noise band, equal in spectral level to the first. The noises were mixed to form notches of various widths centered on the signal frequency. Minimimum signal levels required for 79.4 percent correcr: uetection were measured using an adaptive, two-alternative forced choice procedure. The subject was instructed not to m v e the head, and the chin was positioned in a chin rest.
三维(3-D)听觉显示可以提高操作员在各种应用中的表现，包括声纳(Doll, Hanna, and Ruissotti, in press)，飞机驾驶舱的听觉警告(Doll等)。大多数预期的3-D听觉显示应用涉及同时呈现来自不同方向的多个信号。一个潜在的问题是，同时或紧密发出的信号可能会相互掩盖。同时发出的声音相互掩盖的程度应取决于它们的频谱相似性和它们的源在空间中的位置有多近。众所周知，当掩蔽器和信号不占用相同的临界频带时，掩蔽会大大减少(例如，Durlach & Colburn, 1978)。自由场掩蔽的研究表明，掩蔽器的有效性随着它与信号在空间上的分离而降低。然而，在确定3-0听觉显示信号的可检测性时，频谱和空间相似性权衡的程度是未知的。设计有效的3-D显示器需要这些信息。本研究的目的是确定信号和信号的频谱和空间相似性如何相互作用，以确定三维听觉显示中信号的可探测性。在自由场(即真实的三维听觉显示)中，从具有不同空间间隔(0度、20度和40度)的扬声器中呈现音调信号和陷波噪声掩蔽器。扬声器被布置成一个水平圆弧，距离听众10英尺，在耳朵1 - 5 - 1处。通过改变一个噪声带的低通截止和另一个独立噪声带的高通截止，在光谱电平上与第一个噪声带相等，可以操纵掩蔽器和信号的频谱相似性。这些噪声被混合成以信号频率为中心的不同宽度的凹痕。79.4%正确率所需的最小信号电平:使用自适应，两种选择的强制选择程序进行测量。受试者被指示不要碰头部，下巴放在下巴托上。
{"title":"Masking In Three-dimensional Auditory Displays II: Effects Of Spatial And Spectral Simil2irity","authors":"Theodore J Doll, Thomas E Hanna","doi":"10.1109/ASPAA.1991.634102","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634102","url":null,"abstract":"It has been suggested that three dimensional (3-D) auditory displays could enhance operator performance in a wide variety of applications, including sonar (Doll, Hanna, and Ruissotti, in press), auditory warnings in aircraft cockpits (Doll et al., Most of the anticipated applications of 3-D auditory dislplays involve simultaneous presentation of multiple signals from different directions. A potential problem is that signals that are sounded simultaneously or closely in time may mask one-another. The extent to which simultaneous sounds mask one-(another should depend both upon their spectral similarity and how closely their sources are positioned in space. It is well esta'blished that masking is greatly reduced when the masker and signal do not occupy the same critical band (e.g., Durlach & Colburn, 1978). Studies of free-field masking show that the effectiveness of a masker decreases as it is separated in space from the signal However, the extent to which spectral and spatial similarity trade-off in determining the detectability of signals i n 3-0 auditory displays is unknom. This information i.s needed to design effective 3-D displays. The purpose of this research was to deternine how the spectral and spatial similarity of signals arid naskers interact to determine the deteczability of s i p a i s in 3-D audicory displays. A tonal signal and a **notchedtg noise masker were presented from loudspeakers with various spatial separations (0, 20, and 40 degrees) in a free field (i.e. , a *treal*t 3-D auditory display). The loudspeakers were arranged in a horizontal circular arc 10 ft. from the listener at ear l e v e l. The spectral similarity of the masker and signal were manipulated by varying the low-pass cutoff of one noise band and the high-pass cutoff of another, independent noise band, equal in spectral level to the first. The noises were mixed to form notches of various widths centered on the signal frequency. Minimimum signal levels required for 79.4 percent correcr: uetection were measured using an adaptive, two-alternative forced choice procedure. The subject was instructed not to m v e the head, and the chin was positioned in a chin rest.","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":"130947080","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.1991.634122
A. Ganeshkumar, J. Hammond, C. G. Rice
Our approach is based on enhancing the Short Time Spectral Amplitude (STSA) of degraded speech using the spectral subtraction algorithm. The use of spectral subtraction to enhance speech has been studied quite extensively in the past [1,2]. These studies have generally shown an increase in speech quality but the gain in intelligibility has been insignificant. The lack of improvement in intelligibility can be atmbiited to two main factors. The first being that since all previous work on the application of spectral subtraction algorithm have been confined to single input systems, the noise short time spectrum can only be estimated during non-speech activity periods. This approach not only requires accurate speechhion-speech activity detection a difficult task, particularly at low signal to noise ratiosbut also requires the noise to be sufficiently stationary for the estimate to be used during the subsequent speech period. The second factor for the lack of improvement in intelligibility is due to the annoying 'musical' type of residual noise introduced by spectral subtraction processing. This residual noise may distract the listener from the speech.
{"title":"Speech Enhancement For Hearing Aids Using A Microphone Array","authors":"A. Ganeshkumar, J. Hammond, C. G. Rice","doi":"10.1109/ASPAA.1991.634122","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634122","url":null,"abstract":"Our approach is based on enhancing the Short Time Spectral Amplitude (STSA) of degraded speech using the spectral subtraction algorithm. The use of spectral subtraction to enhance speech has been studied quite extensively in the past [1,2]. These studies have generally shown an increase in speech quality but the gain in intelligibility has been insignificant. The lack of improvement in intelligibility can be atmbiited to two main factors. The first being that since all previous work on the application of spectral subtraction algorithm have been confined to single input systems, the noise short time spectrum can only be estimated during non-speech activity periods. This approach not only requires accurate speechhion-speech activity detection a difficult task, particularly at low signal to noise ratiosbut also requires the noise to be sufficiently stationary for the estimate to be used during the subsequent speech period. The second factor for the lack of improvement in intelligibility is due to the annoying 'musical' type of residual noise introduced by spectral subtraction processing. This residual noise may distract the listener from the speech.","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":"123062308","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.1991.634131
D. Rossum
In the &sign of VLSI circuits to implement digital filters for electronic music purposes, we have found it useful to encode the filter coefficients. Such encoding offers three advantages. First, the encoding can be made to correspond more properly to the "natural" perceptual units of audio. While these are most accurately the "bark" for frequency and the "sone" for loudness, a good working approximation is decibels and musical octaves respectively. Secondly, our encoding scheme allows for partial decoupling of the pole radius and angle, providing superior interpolation characteristics when the coefficients are dynamically swept. Thirdly, and perhaps most importantly, appropriate encoding of the coefficients can save substantial amounts of on-chip memory. While audio filter coefficients typically require twenty or more bits, we have found adequate coverage at as few as eight bits, allowing for a much more cost effective custom hardware implementation when many coefficients are required. We have named the resulting patented encoding scheme "ARh4Adillo." Our implementation of digital audio filters is based on the canonical second order section whose transfer function should be familiar to all: 1*-1*-2 H(Z) = +blz-1+b,z-2 [I1 While dealing with poles and feedback (bn) coefficients, the comments herein apply as well to zeroes and feedforward coefficients (an/@) when the gain (a@ is separated as shown above. Noting that the height of a resonant peak in the magnitude response produced by a pole is approximately inversely proportional to the distance from the pole to the unit circle, we can relate the height p of this resonant peak in dB to the pole radius R: 1 1-R
在VLSI电路的&符号中实现用于电子音乐的数字滤波器，我们发现对滤波器系数进行编码是有用的。这样的编码提供了三个优点。首先，编码可以更恰当地对应于音频的“自然”感知单元。虽然用“bark”来表示频率最准确，用“sone”来表示响度最准确，但一个很好的近似方法是分别用分贝和八度。其次，我们的编码方案允许极点半径和角度的部分解耦，当系数被动态扫描时提供优越的插值特性。第三，也许是最重要的，系数的适当编码可以节省大量的片上存储器。虽然音频滤波器系数通常需要20位或更多位，但我们发现只需8位就足够覆盖，当需要许多系数时，允许更具成本效益的定制硬件实现。我们将由此产生的专利编码方案命名为“ARh4Adillo”。我们的数字音频滤波器的实现是基于规范的二阶部分，其传递函数应该是所有人都熟悉的:1*-1*-2 H(Z) = +blz-1+b, Z -2 [I1]在处理极点和反馈(bn)系数时，此处的注释也适用于零点和前馈系数(an/@)，当增益(a@)如上所示分离时。注意到极点产生的振幅响应中谐振峰的高度与极点到单位圆的距离近似成反比，我们可以将这个谐振峰的高度p(以dB为单位)与极点半径R联系起来:1 1-R
{"title":"The \"ARMAdillo\" Coefficient Encoding Scheme for Digital Audio Filters","authors":"D. Rossum","doi":"10.1109/ASPAA.1991.634131","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634131","url":null,"abstract":"In the &sign of VLSI circuits to implement digital filters for electronic music purposes, we have found it useful to encode the filter coefficients. Such encoding offers three advantages. First, the encoding can be made to correspond more properly to the \"natural\" perceptual units of audio. While these are most accurately the \"bark\" for frequency and the \"sone\" for loudness, a good working approximation is decibels and musical octaves respectively. Secondly, our encoding scheme allows for partial decoupling of the pole radius and angle, providing superior interpolation characteristics when the coefficients are dynamically swept. Thirdly, and perhaps most importantly, appropriate encoding of the coefficients can save substantial amounts of on-chip memory. While audio filter coefficients typically require twenty or more bits, we have found adequate coverage at as few as eight bits, allowing for a much more cost effective custom hardware implementation when many coefficients are required. We have named the resulting patented encoding scheme \"ARh4Adillo.\" Our implementation of digital audio filters is based on the canonical second order section whose transfer function should be familiar to all: 1*-1*-2 H(Z) = +blz-1+b,z-2 [I1 While dealing with poles and feedback (bn) coefficients, the comments herein apply as well to zeroes and feedforward coefficients (an/@) when the gain (a@ is separated as shown above. Noting that the height of a resonant peak in the magnitude response produced by a pole is approximately inversely proportional to the distance from the pole to the unit circle, we can relate the height p of this resonant peak in dB to the pole radius R: 1 1-R","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":"115024907","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.1991.634098
S. Foster, E. Wenzel, R. M. Tayior
This paper describes some recent efforts to "render" the complex acoustic field experienced by a listener within an environment. It represents an extension of earlier attempts to synthesize externalized, threedimensional sound cues over headphones using a very high-speed, signal processor, the Convolvotron (Wenzel, et al., 1988). The synthesis technique involves the digital generation of stimuli using HeadRelated Transfer Functions (HRTFs) measured in the ear canals of individual subjects for a large number of equidistant kcations in an anechoic chamber (Wightman & Kistler, 1989). The advantage of this technique is that it preserves the complex pattern of interaural differences over the entire spectrum of the stimulus, thus capturing the effects of filtering by the pinnae, head, shoulders, and torso.
{"title":"Real Time Synthesis of Complex Acoustic Environments","authors":"S. Foster, E. Wenzel, R. M. Tayior","doi":"10.1109/ASPAA.1991.634098","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634098","url":null,"abstract":"This paper describes some recent efforts to \"render\" the complex acoustic field experienced by a listener within an environment. It represents an extension of earlier attempts to synthesize externalized, threedimensional sound cues over headphones using a very high-speed, signal processor, the Convolvotron (Wenzel, et al., 1988). The synthesis technique involves the digital generation of stimuli using HeadRelated Transfer Functions (HRTFs) measured in the ear canals of individual subjects for a large number of equidistant kcations in an anechoic chamber (Wightman & Kistler, 1989). The advantage of this technique is that it preserves the complex pattern of interaural differences over the entire spectrum of the stimulus, thus capturing the effects of filtering by the pinnae, head, shoulders, and torso.","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":"125684487","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.1991.634151
P. Depalle, D. Matignon, X. Rodet
In this paper, we present an approach to sound synthesis which mes to unify the two current approaches, one that we call the signal approach and the other that we call the physical approach. These two approaches have their own advantages and drawbacks. 1. The signal approach inherits the whole set of signal processing techniques. It is based on the use of fairly general production models, the internal structure of which is not precisely defined. The input variables to the model are called parameters. The process of synthesizing a sound consists of finding the time varying values of the parameters. In general, there exist analysis techniques to determine parameter values from natural sounds (e.g. FFT ifor additive synthesis, LPC for source filter models). One of the drawbacks to this approach is the difficulty in determining the parameter values of a signal whose Characteristics vary rapidly. It is also difficult to control the model for certain sound effects since there is no internal description. 2. The physical approach consists of an explicit simulation of the physical system which produces the sound. In this case the internal description is precisely defined. Synthesis is accomplished by finding the numeric solution to the model equation. The control parameters directly correspond to the physical parameters of the system. The sound produced by such models are of great quality. The drawback to this synthesis method is that the modd equations are determined from a dePailed physical analysis of the insuument and that the parameters have to be obtained from physical measurements which are often long and complex to realise. To take advantage of the positive aspects of the preceding approaches, we explore a third approach. On the one hand it takes advantage of a precise description of the internal structure of a physical system. On the other hand, it determines certain parameter values by analyzing sounds produced by the system. Our new approach is based on the state variable description of physical systems. This formalism is largely used in process control theory. Kalmari filtering is one of the techniques that we use in order to obtain the parameter values that conool ihe model. We have applied this formalism to build a model of connected acoustic tubes. We have developped an algorithm for recursive consrmction of a state variable model given the structure of the system. Such a model can be excited by non linear systems to …
{"title":"State Variable Models For Sound Synthesis","authors":"P. Depalle, D. Matignon, X. Rodet","doi":"10.1109/ASPAA.1991.634151","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634151","url":null,"abstract":"In this paper, we present an approach to sound synthesis which mes to unify the two current approaches, one that we call the signal approach and the other that we call the physical approach. These two approaches have their own advantages and drawbacks. 1. The signal approach inherits the whole set of signal processing techniques. It is based on the use of fairly general production models, the internal structure of which is not precisely defined. The input variables to the model are called parameters. The process of synthesizing a sound consists of finding the time varying values of the parameters. In general, there exist analysis techniques to determine parameter values from natural sounds (e.g. FFT ifor additive synthesis, LPC for source filter models). One of the drawbacks to this approach is the difficulty in determining the parameter values of a signal whose Characteristics vary rapidly. It is also difficult to control the model for certain sound effects since there is no internal description. 2. The physical approach consists of an explicit simulation of the physical system which produces the sound. In this case the internal description is precisely defined. Synthesis is accomplished by finding the numeric solution to the model equation. The control parameters directly correspond to the physical parameters of the system. The sound produced by such models are of great quality. The drawback to this synthesis method is that the modd equations are determined from a dePailed physical analysis of the insuument and that the parameters have to be obtained from physical measurements which are often long and complex to realise. To take advantage of the positive aspects of the preceding approaches, we explore a third approach. On the one hand it takes advantage of a precise description of the internal structure of a physical system. On the other hand, it determines certain parameter values by analyzing sounds produced by the system. Our new approach is based on the state variable description of physical systems. This formalism is largely used in process control theory. Kalmari filtering is one of the techniques that we use in order to obtain the parameter values that conool ihe model. We have applied this formalism to build a model of connected acoustic tubes. We have developped an algorithm for recursive consrmction of a state variable model given the structure of the system. Such a model can be excited by non linear systems to …","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":"133228704","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.1991.634096
O. Ghitza
A long standing question that arises when studying a particular auditory model is how to evaluate its performance. More precisely, it is of interest to evaluate to what extent the model-representation can describe the actual human internal representation. In this study we address this question in the context of speech perception. That is, given a speech representation based on the auditory system, to what extent can it preserve phonetic information that is perceptually relevant? To answer this question, a diagnostic system has been developed that simulates the psychophysical procedure used in the standard Diagnostic-Rhyme Test (DRT, Voiers, 1983). In the psychophysical procedure the subject has all the cognitive information needed for the discrimination task a priori. Hence, errors in discrimination are due mainly to inaccuracies in the auditory representation of the stimulus. In the simulation, the human observer is replaced by an array of recognizers, one for each pair of words in tlhe DRT database. An effort has been made to keep the errors due to the "observer" to a minimum, so that the overall detected errors are due mainly to inaccuracies in the auditory model representation. This effort includes a careful design of the recognizer (i.e, using an HMM with time-varying states, Ghitza and Sondhi, 1990) and the use of a speaker-dependent DRT simulation. To demonstrate the power of the suggested evaluation method, we considered the behavior of two speech analysis methods, the Fourier power spectrum and a representation based on the auditory syslem (the EIH model, Ghitza, 1988), in, quiet and in a noisy environment. The results were compared with psychophysical results for the same database. The results show that the overall number of errors made by the machine (the Fourier power spectrum or the EIK) are far greater than the overall number of errors made by a human, at all noise llevels that were tested. Further, the proposed evaluation method offers a detailed picture of the error distribution among the selected phonetic features. It shows that the errors made by the human listener sue distributed in a different way compared to the errors made by the machines, and that the distributions of errors made by the two analyzers are also quiet different from each other.
{"title":"An Acoustic-Phonetic Diagnostic Tool for the Evaluation of Auditory Models","authors":"O. Ghitza","doi":"10.1109/ASPAA.1991.634096","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634096","url":null,"abstract":"A long standing question that arises when studying a particular auditory model is how to evaluate its performance. More precisely, it is of interest to evaluate to what extent the model-representation can describe the actual human internal representation. In this study we address this question in the context of speech perception. That is, given a speech representation based on the auditory system, to what extent can it preserve phonetic information that is perceptually relevant? To answer this question, a diagnostic system has been developed that simulates the psychophysical procedure used in the standard Diagnostic-Rhyme Test (DRT, Voiers, 1983). In the psychophysical procedure the subject has all the cognitive information needed for the discrimination task a priori. Hence, errors in discrimination are due mainly to inaccuracies in the auditory representation of the stimulus. In the simulation, the human observer is replaced by an array of recognizers, one for each pair of words in tlhe DRT database. An effort has been made to keep the errors due to the \"observer\" to a minimum, so that the overall detected errors are due mainly to inaccuracies in the auditory model representation. This effort includes a careful design of the recognizer (i.e, using an HMM with time-varying states, Ghitza and Sondhi, 1990) and the use of a speaker-dependent DRT simulation. To demonstrate the power of the suggested evaluation method, we considered the behavior of two speech analysis methods, the Fourier power spectrum and a representation based on the auditory syslem (the EIH model, Ghitza, 1988), in, quiet and in a noisy environment. The results were compared with psychophysical results for the same database. The results show that the overall number of errors made by the machine (the Fourier power spectrum or the EIK) are far greater than the overall number of errors made by a human, at all noise llevels that were tested. Further, the proposed evaluation method offers a detailed picture of the error distribution among the selected phonetic features. It shows that the errors made by the human listener sue distributed in a different way compared to the errors made by the machines, and that the distributions of errors made by the two analyzers are also quiet different from each other.","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":"127701484","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.1991.634148
R. Maher
Amplitude beating between closely spaced frequency components is a well-known effect in musical acoustics, psychoacoustics, and other fields [l-31. Depending on the musical context and the personal preference of the listener, the presence of amplitude lbeating can either be an undesirable artifact of the limited frequency resolution of the human hearing apparatus, or a pleasant quality that adds timbral variety to an ensemble performance. In either case it would be useful to be able to control the extent of interharmonic amplitude beating in some convenient manner. The increased use of digital computing systems in music synthesis and post-production opens up many new avenues for innovative digital signal processing. This paper extends the repertoire of digital audio signal processing methods to include direct control over amplitude beating in complex audio signals due to interaction among spectral components of simultaneous musical voices. Applications of this technique include 1) discriminability improvement for weak or easily masked musical voices in complex sonic textures, and 2) a1 teration of the consonance/dissonance relationship of musical intervals and chords to retain the advantages of equal tempered tuning (for example, modulation between keys) while reducing the effects of out of tune pnrtials. Overview As previously reported [4], one means to reduce amplitude beating during additive mixing operations is to perform a time variant spectral analysis on the signals to be mixed, identify the presence of closely spaced frequency components, and selectively attenuate those components which will give rise to amplitude beats. A convenient formulation for this procedure was found to be the sinewave model of McAulay and Quatieri [5]. This approach can be described as excf usion filtering, where one of the signals to be mixed is used to design a time varying comb-like filter to exclude competing spectral energy from the other signals. The amplitude beating among closely spaced partials can also be increased to improve the detectability of a relatively ,weak musical voice in the presence of a complex background ensemble. The increased beatiing is accomplished by using time variant sinusoidal analysis to identify spectral collisions arnong the competing musical voices and then to increase the amplitude of the beating components. This technique is particularly useful when the weak voice has spectral energy in a confined range which overlaps the background material, e.g., a solo clarinet with string ensemble accompaniment. While simply boosting the level of the weak voice can improve its detectability, the combination of increased level a n d enhancemen't of interharmonic beating can increase the perceived separation between the weak signal and its competition. In other words, the presence of the weak voice is cued by its effect upon the other voices in the ensemble.
{"title":"Control Of Interharmonic In Polyphonic Music","authors":"R. Maher","doi":"10.1109/ASPAA.1991.634148","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634148","url":null,"abstract":"Amplitude beating between closely spaced frequency components is a well-known effect in musical acoustics, psychoacoustics, and other fields [l-31. Depending on the musical context and the personal preference of the listener, the presence of amplitude lbeating can either be an undesirable artifact of the limited frequency resolution of the human hearing apparatus, or a pleasant quality that adds timbral variety to an ensemble performance. In either case it would be useful to be able to control the extent of interharmonic amplitude beating in some convenient manner. The increased use of digital computing systems in music synthesis and post-production opens up many new avenues for innovative digital signal processing. This paper extends the repertoire of digital audio signal processing methods to include direct control over amplitude beating in complex audio signals due to interaction among spectral components of simultaneous musical voices. Applications of this technique include 1) discriminability improvement for weak or easily masked musical voices in complex sonic textures, and 2) a1 teration of the consonance/dissonance relationship of musical intervals and chords to retain the advantages of equal tempered tuning (for example, modulation between keys) while reducing the effects of out of tune pnrtials. Overview As previously reported [4], one means to reduce amplitude beating during additive mixing operations is to perform a time variant spectral analysis on the signals to be mixed, identify the presence of closely spaced frequency components, and selectively attenuate those components which will give rise to amplitude beats. A convenient formulation for this procedure was found to be the sinewave model of McAulay and Quatieri [5]. This approach can be described as excf usion filtering, where one of the signals to be mixed is used to design a time varying comb-like filter to exclude competing spectral energy from the other signals. The amplitude beating among closely spaced partials can also be increased to improve the detectability of a relatively ,weak musical voice in the presence of a complex background ensemble. The increased beatiing is accomplished by using time variant sinusoidal analysis to identify spectral collisions arnong the competing musical voices and then to increase the amplitude of the beating components. This technique is particularly useful when the weak voice has spectral energy in a confined range which overlaps the background material, e.g., a solo clarinet with string ensemble accompaniment. While simply boosting the level of the weak voice can improve its detectability, the combination of increased level a n d enhancemen't of interharmonic beating can increase the perceived separation between the weak signal and its competition. In other words, the presence of the weak voice is cued by its effect upon the other voices in the ensemble.","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":"131586005","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.1991.634126
J. Walliker, J. Daley, A. Faulkner, I. Howard
Many people with profound hearing impairment, while able to detect amplified sound are often unable to make sense of what they hear. Conventional hearing aids which amplify, filter and compress the speech signal are of little use to them. It has been demonstrated that some profoundly deaf listeners are able to make better use of speech features such as voice fundamental frequency (Fx) and frication when they are presented in a simplified form matched to their residual hearing than when conventionally presented.
{"title":"Feature extracting hearing aids for the profoundly deaf using a neural network implemented on a TMS320C51 digital signal processor","authors":"J. Walliker, J. Daley, A. Faulkner, I. Howard","doi":"10.1109/ASPAA.1991.634126","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634126","url":null,"abstract":"Many people with profound hearing impairment, while able to detect amplified sound are often unable to make sense of what they hear. Conventional hearing aids which amplify, filter and compress the speech signal are of little use to them. It has been demonstrated that some profoundly deaf listeners are able to make better use of speech features such as voice fundamental frequency (Fx) and frication when they are presented in a simplified form matched to their residual hearing than when conventionally presented.","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":"129589245","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.1991.634088
B. Delgutte, P. Cariani
In principle, speech and audio coding systems can be evaluated by comparing the responses of a model of auditory processing to the original and the coded signals, providiing that the model responses includes all perceptually-relevant features of the signal, while decreasing signal redundancy. In practice, present knowledge of auditory physiology is incomplete, so that it is difficult to decide which aspects of auditory processing the model should attempt to simulate. This; talk will address three fundamental issues in auditory modeling that are important for the design of irnproved coding systems.
{"title":"Fundamental issues in auditory modeling","authors":"B. Delgutte, P. Cariani","doi":"10.1109/ASPAA.1991.634088","DOIUrl":"https://doi.org/10.1109/ASPAA.1991.634088","url":null,"abstract":"In principle, speech and audio coding systems can be evaluated by comparing the responses of a model of auditory processing to the original and the coded signals, providiing that the model responses includes all perceptually-relevant features of the signal, while decreasing signal redundancy. In practice, present knowledge of auditory physiology is incomplete, so that it is difficult to decide which aspects of auditory processing the model should attempt to simulate. This; talk will address three fundamental issues in auditory modeling that are important for the design of irnproved coding systems.","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":"127231441","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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