{"title":"Analysis of ill-conditioning of multi-channel deconvolution problems","authors":"O. Kirkeby, P. Rubak, A. Farina","doi":"10.1109/ASPAA.1999.810873","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810873","url":null,"abstract":"Deconvolution of single- and multichannel systems is often an ill-conditioned problem whose exact solution boosts certain frequency bands excessively. Frequency-dependent regularisation can used to prevent this by attenuating sharp peaks in the magnitude response of the optimal filters. A z-domain analysis demonstrates that frequency-dependent regularisation works by pushing the poles of an ideal optimal solution away from the unit circle.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"134 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133919010","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":"Reduced-rank modeling of head-related impulse responses using subset selection","authors":"W. G. Gardner","doi":"10.1109/ASPAA.1999.810878","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810878","url":null,"abstract":"This paper describes the subset selection method of modeling head-related impulse responses. The subset method is mathematical technique for choosing a set of basis functions directly from the data set that is being modeled. The subset method is compared to principal component analysis and the virtual speaker method.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"18 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117287912","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":"Analysis and enhancement of locally harmonic signals using adaptive multi-kernel methods","authors":"S. Dubost, O. Cappé","doi":"10.1109/ASPAA.1999.810875","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810875","url":null,"abstract":"This paper is concerned with the analysis/synthesis and enhancement of signals that can be efficiently modeled as \"quasi-harmonic\". We first provide a formal definition of quasi-harmonic signals and show that known estimation methods for such signals can be interpreted in a non-parametric local approximation framework. We then investigate the possibility of using \"adaptive multi-kernel\" estimation methods for which the duration of the analysis frame is tuned independently for each harmonic depending on the local behavior of the signal.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"52 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125041898","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":"\"Preference space\" for parameters of two-channel speech compression","authors":"W. S. Woods, R.A. Frush, D. V. Van Tasell","doi":"10.1109/ASPAA.1999.810893","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810893","url":null,"abstract":"Speech processed with amplitude compression can maintain high levels of intelligibility over a wide range of compression parameter values. The perceived quality of the speech, however, varies widely over this parameter range, especially in the presence of low-level background noise. The present work provides data showing that both normal-hearing listeners and listeners with impairments prefer low compression ratios and long time constants in two-channel compression processing. In addition, we describe a quantitative model linking the release time and compression ratio of the two-channel compression system to the preference judgments of normal-hearing and hearing impaired listeners.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"24 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124599686","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":"Broadband beamforming optimization for speech enhancement in noisy environments","authors":"Matti Kujula, Matti Hdmalainen","doi":"10.1109/ASPAA.1999.810839","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810839","url":null,"abstract":"We have developed a method to optimize the directional sensitivity of a filter-and-sum beamformer. The directivity of the broadband microphone array is optimized by adjusting the spatial transducer positions and the impulse response of the beamformer. We focus on the optimization of 1-dimensional arrays consisting of M omnidirectional microphones and M FIR filters each of length L. The signal sources are assumed to be point sources evenly distributed over a sphere of radius r or over a set of concentric spheres. The target beam pattern for optimization is defined in terms of the desired source signal directions (mainlobe) and the angles for background noise attenuation.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"37 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130564994","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":"An investigation of the impact of torsion waves and friction characteristics on the playability of virtual bowed strings","authors":"S. Serafin, J.O. Smith, J. Woodhouse","doi":"10.1109/ASPAA.1999.810856","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810856","url":null,"abstract":"\"Playability\" is measured for variations in a bowed string simulation model. The variations studied are (1) the effect of torsion waves, and (2) the effect of the choice of friction model. It is found that (1) elimination of torsion wave simulation does not degrade playability, and (2) the more recently developed \"plastic\" bowed-string friction model, in which the frictional force is a function of temperature, is more \"playable\" than prior friction models which depend only on relative sliding velocity.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"38 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130940954","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":"Signal selection for the room acoustics measurement","authors":"I. Mateljan","doi":"10.1109/ASPAA.1999.810884","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810884","url":null,"abstract":"The paper describes a class of random phase multisine signals and their use in a single channel impulse response measurement system. The system is compared with a maximum length sequence driven system. The influence of measuring system nonlinearities and noise on the impulse response estimation is analyzed. The experimental work has confirmed the advantage of using the random phase multisine excitation in room acoustics measurements.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"96 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122530579","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":"Low-dimensional audio-rate control of FFT-based processing","authors":"C. Lippe, Zack Settel","doi":"10.1109/ASPAA.1999.810858","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810858","url":null,"abstract":"While the use of the fast Fourier transform (FFT) for signal processing in music applications has been widespread, applications in real-time systems for dynamic spectral transformation have been quite limited. The limitations have been largely due to the amount of computation required for the operations. With faster machines, and with suitable implementation for frequency-domain processing, real-time dynamic control of high-quality spectral processing can be accomplished with great efficiency and a simple approach. This paper describe some previous work in dynamic real-time control of frequency-domain-based signal processing. Since the implementation of the FFT/IFFT is central to the approach and methods discussed, the authors provide a description of this implementation, as well as of the development environment used in their work.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130417673","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":"Pitch estimation using multiple independent time-frequency windows","authors":"Anssi Klapuri","doi":"10.1109/ASPAA.1999.810863","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810863","url":null,"abstract":"A system for the detection of the pitch of musical sounds at a wide pitch range and in diverse conditions is presented. The system is built upon a pitch model that calculates independent pitch estimates in separate time-frequency windows and then combines them to yield a single estimate of the pitch. Both psychoacoustic and computational experiments were carried out to determine the optimal sizes of the elementary windows. The robustness of the system in wide-band additive noise and in the interference of another harmonic sound are demonstrated. An extension of the algorithm to the multi-pitch case is described, and simulation results for two-voice polyphonics are presented.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"39 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128553907","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}
Takanori Nishino, S. Kajita, K. Takeda, F. Itakura
{"title":"Interpolating head related transfer functions in the median plane","authors":"Takanori Nishino, S. Kajita, K. Takeda, F. Itakura","doi":"10.1109/ASPAA.1999.810876","DOIUrl":"https://doi.org/10.1109/ASPAA.1999.810876","url":null,"abstract":"This paper describes the interpolation of head related transfer functions (HRTFs) for all direction in the median plane. The interpolation of HRTFs enables us to reduce the number of measurements for new user's HRTFs, and also reduce the data of HRTFs in auditory virtual systems. In this paper, a simple linear interpolation method and the spline interpolation method are evaluated and advantages of both methods clarified. In experiments, the interpolation methods are applied to HRTFs measured using a dummy head. The experimental results show that the two methods are comparable in the best case. The resultant minimum spectral distortion is about 2 dB for both methods. The results clarify that the linear interpolation is effective for a set of elevations selected based on the cross correlation and that the spline interpolation is effective at large and equal intervals. These results indicate that HRTFs in the median plane can be interpolated by the methods.","PeriodicalId":229733,"journal":{"name":"Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1999-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125719971","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}