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Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics最新文献

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A spherical basis function neural network for pole-zero modeling of head-related transfer functions 用于头部相关传递函数极点-零点建模的球面基函数神经网络
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482966
Rick L. Jenison
{"title":"A spherical basis function neural network for pole-zero modeling of head-related transfer functions","authors":"Rick L. Jenison","doi":"10.1109/ASPAA.1995.482966","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482966","url":null,"abstract":"This paper describes a neural network for approximating the parameters of a pole-zero model of the head-related transfer function (HRTF). The von Mises basis function (VMBF) is described whose response depends on spherical rather than Cartesian input coordinates. The VMBF neural network is ideally suited to the problem of learning a continuous mapping from spherical coordinates to acoustic parameters that specify sound source direction. A method for computing the common poles of a set of HRTFs is also discussed.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128152175","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}
引用次数: 19
Using a fast-recursive-least-squared algorithm in a feedback-controller 在反馈控制器中使用快速递归最小二乘算法
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482948
Th. Auspitzer, Guicking, Elliott
{"title":"Using a fast-recursive-least-squared algorithm in a feedback-controller","authors":"Th. Auspitzer, Guicking, Elliott","doi":"10.1109/ASPAA.1995.482948","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482948","url":null,"abstract":"In active noise control applications (ANC) recursive least-squared algorithms have become more and more common. The reason is that their convergence behavior is quite independent of the statistics of the input sound signal. Using such algorithms with digital finite impulse response filters, FIR, of an order N they show a typical convergence within N samples. There exist several algorithms for FIR-filters which are often called fast recursive least-squared (FRLS) or as fast transversal filters (FTF) in the related literature. The controller is usually built as a feedforward controller when these fast algorithms are used. The crucial integration of the secondary or error path is not done in the same way as for the well known LMS-algorithm. For the LMS usually the filtered-X-LMS is used. But this integration scheme does not work with those fast algorithms. Integration methods like the FASPIS-scheme are used instead. We describe a way to use these efficient algorithms in a adaptive feedback control structure.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126411849","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}
引用次数: 9
Second order Volterra inverses for compensation of loudspeaker nonlinearity 扬声器非线性补偿的二阶Volterra逆
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482982
H. Schurer, C. Slump, O. E. Herrmann
{"title":"Second order Volterra inverses for compensation of loudspeaker nonlinearity","authors":"H. Schurer, C. Slump, O. E. Herrmann","doi":"10.1109/ASPAA.1995.482982","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482982","url":null,"abstract":"High quality sound reproduction by loudspeakers is increasingly problematic if the dimensions of the loudspeaker decrease. To produce enough power, large diaphragm excursions are needed which give rise to significant distortions especially at very low frequencies. Instead of improving the mechanical construction of the transducer we apply a feedforward nonlinear digital inverse circuit. Results of two 2nd order Volterra compensators show a significant reduction of the second order harmonics, leaving higher order distortions unchanged. The structure of the realization influences the performance considerably. Two realization structures are considered, and the error caused by the differentiators in the output of the compensators are compared. Both algorithms are implemented in real-time on a digital signal processor (DSP) for on-line testing with the transducer.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"73 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132100526","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}
引用次数: 16
Observations on auditory excitation and masking patterns 听觉兴奋和掩蔽模式的观察
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482992
S. Voran
{"title":"Observations on auditory excitation and masking patterns","authors":"S. Voran","doi":"10.1109/ASPAA.1995.482992","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482992","url":null,"abstract":"Excitation patterns and masking patterns are used extensively in perceptual audio coders and quality assessment algorithms. Numerous algorithms for calculating these patterns have been proposed. This paper provides comparisons among the patterns generated by several of these algorithms. The comparisons are based on audio program material, rather than tones and noise. Explored areas include synthesis functions, spreading functions, masking indices, tonality measures, and the treatment of the absolute threshold of hearing. Several mathematical relations are provided to characterize observations in these areas. Patterns from simpler algorithms are considered as approximations to patterns from more complex algorithms, and the approximation error is characterized. Results may be useful to those who apply auditory excitation or masking patterns in their work.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"53 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122698295","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}
引用次数: 12
Fixed filter implementation of feedback cancellation for in-the-ear hearing aids 固定滤波器实现耳内式助听器的反馈取消
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482904
B. Woodruff, D. Preves
{"title":"Fixed filter implementation of feedback cancellation for in-the-ear hearing aids","authors":"B. Woodruff, D. Preves","doi":"10.1109/ASPAA.1995.482904","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482904","url":null,"abstract":"Acoustic feedback oscillation in hearing aids is a problem which often prevents an adequate amount of gain from being realized for some hearing aid users. A study is presented in which a feedback cancellation technique is applied to hearing aids in 13 ears. The technique consists of modeling the feedback path around the amplifier and duplicating it with an FIR (finite impulse response) filter. The output of this filter is inverted and summed with the input signal at the input to the amplifier, canceling the signal from the feedback path around the amplifier. This technique is applied in its most optimized form and also in several variations of a simplified form. The optimized form employs a long floating point FIR filter which is generated from the impulse response of each specific subject's feedback path around the amplifier. These feedback paths include the microphone and receiver responses in a vented in-the-ear (ITE) earmold. Comparisons between feedback path transfer functions are shown to reveal that the phase responses are quite similar, but that the magnitude responses are not. An FIR filter is constructed which is an average of the 13 measured impulse responses. This FIR is implemented in three forms: a long floating point FIR filter, a short floating point FIR filter, and a short fixed point FIR filter. All three are shown to be effective in reducing feedback, provided the overall gain of the filter may be increased or decreased in order to optimally suit the magnitude response of the filter to the needs of each person's ear. Using this technique a fixed filter may be implemented in ITE hearing aids for all hearing aid wearers to allow more gain without feedback oscillation.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"256 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115870294","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}
引用次数: 3
Adaptive minimisation of the maximum error signal in an active control system 主动控制系统中最大误差信号的自适应最小化
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482946
A. González, S. Elliott
{"title":"Adaptive minimisation of the maximum error signal in an active control system","authors":"A. González, S. Elliott","doi":"10.1109/ASPAA.1995.482946","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482946","url":null,"abstract":"Most active noise control systems with multiple error sensors minimise the sum of the modulus squared outputs of these sensors. The purpose of this paper is to present an adaptive algorithm which minimises an alternative cost function which, in the limit, is equal to the maximum modulus squared value of all the error sensors. We investigate the physical consequences of minimise the maximum modulus squared output in this set to achieve noise reductions. The work is focused on an algorithm called the minimax algorithm. This algorithm has been studied in depth for an active control problem concerning one primary source, one secondary source and multiple error signals. The analysis covers steady state performance and convergence properties. In order to compare the proposed algorithm with the classical least squares solutions, a linear model of a one-dimensional acoustic system has been used, working at a single frequency.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127820622","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}
引用次数: 2
Wavelet transform and frequency domain kurtosis: application to assessment of hearing hazard from noise exposure 小波变换和频域峰度分析在噪声听力危害评价中的应用
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482903
S. Lei, R. Hamernik
{"title":"Wavelet transform and frequency domain kurtosis: application to assessment of hearing hazard from noise exposure","authors":"S. Lei, R. Hamernik","doi":"10.1109/ASPAA.1995.482903","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482903","url":null,"abstract":"Research on the effects of noise on the auditory system has shown that the frequency domain kurtosis (FDK) statistic could be a useful metric in quantifying a noise environment for hearing conservation purposes. Joint time-frequency analytical methods used in signal processing may be of use in the extraction of such information from an exposure environment characterized by nonstationary signals. This paper shows that the FDK metric can be efficiently computed by applying the wavelet transform to the analysis of an acoustic signal.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128791461","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}
引用次数: 5
Some methods for detection and interpolation of impulsive distortions in old audio recordings 旧录音中脉冲失真的检测和插值方法
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482976
A. Czyżewski
{"title":"Some methods for detection and interpolation of impulsive distortions in old audio recordings","authors":"A. Czyżewski","doi":"10.1109/ASPAA.1995.482976","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482976","url":null,"abstract":"Some techniques for the detection of impulsive distortions such as FFT- and wavelet-based ones were studied. A neural network algorithm was employed to the task of detection of clicks. Some methods for the interpolation of lost samples such as nonlinear interpolation and interpolation utilizing linear predictive coding were investigated. The elaborated neural network was trained in such a way that it is performing some kind of nonlinear predictive coding of the signal, providing a valuable tool for the retrieval of samples lost inside the clicks. Some new algorithms for the interpolation of lost samples is presented.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"29 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128370349","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}
引用次数: 13
Multiple beam broadband beamforming: filter design and real-time implementation 多波束宽带波束形成:滤波器设计和实时实现
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482984
S. Van Gerven, Dirk Van Compernolle, P. Wauters, W. Verstraeten, K. Eneman, K. Delaet
{"title":"Multiple beam broadband beamforming: filter design and real-time implementation","authors":"S. Van Gerven, Dirk Van Compernolle, P. Wauters, W. Verstraeten, K. Eneman, K. Delaet","doi":"10.1109/ASPAA.1995.482984","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482984","url":null,"abstract":"In this paper the design and implementation of a truly broadband beamformer is described. Using only a limited number of microphones a beamformer has been developed with a reasonably homogeneous beampattern over the entire speech bandwidth. The solution is obtained as the result of a filter-and-sum operation. Beams are steered towards different listening-directions and for each direction a set of spatio-temporal filters had to be designed. All beams operate in parallel and a beam-selection procedure picks out the beam with the most prominent speech signal. The complete system has been implemented in real-time using 1 MOTOROLA-56001 and 4 TMS-320C40 processors.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"34 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129252459","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}
引用次数: 14
Time-scale modification with inconsistent constraints 具有不一致约束的时间尺度修改
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.483004
T. Quatieri, T. E. Hanna
{"title":"Time-scale modification with inconsistent constraints","authors":"T. Quatieri, T. E. Hanna","doi":"10.1109/ASPAA.1995.483004","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.483004","url":null,"abstract":"A set theoretic estimation approach is introduced for time-scale modification of complex acoustic signals. The method determines a signal that meets, in a least-squared error sense, desired temporal and spectral envelope constraints that are inconsistent. These constraints are generalized within the set theoretic framework to include other signal characteristics such as instantaneous frequency and group delay. The approach can enhance acoustic signals consisting of closely-spaced sequential time components, and is applicable to biological, underwater, and music sound processing.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"26 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129566074","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}
引用次数: 5
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