{"title":"Optimizing the synthesis filter bank in audio coding for minimum distortion using a frequency weighted psychoacoustic criterion","authors":"K. Gosse, O. Pothier, P. Duhamel","doi":"10.1109/ASPAA.1995.482988","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482988","url":null,"abstract":"This paper shows the improvement brought in filter-bank based audio coding schemes by relaxing the constraints of perfect reconstruction or maximum frequency selectivity on the synthesis filter bank: here, the coding scheme is optimized according to a frequency weighted psychoacoustic criterion. This improvement is then evaluated by means of rate/distortion curves (the weighted criterion being the distortion measure). This approach does not take frequency masking effects into account but is compatible with their use.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"43 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":"126475910","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":"Simulation of teleconferencing environments","authors":"C. Thompson, K. Chandra, V. Mehta","doi":"10.1109/ASPAA.1995.482975","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482975","url":null,"abstract":"A new procedure for evaluating the impulse response from source to receiver location in a room is outlined. The source to receiver impulse response is based on the method of images and is extended to incorporate scattering from objects having high acoustical contrast. It is shown that the impulse response in the presence of scatterers can be computed using Pade approximants to represent the scattered acoustic pressure field.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"1 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":"125859608","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 optimal auditory filter","authors":"I. Toshio","doi":"10.1109/ASPAA.1995.482990","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482990","url":null,"abstract":"The optimality of the peripheral auditory filter is investigated using operator methods applied to a scale representation. A 'gammachirp' function, which consists of a frequency modulated carrier and an envelope of a gamma distribution function, is found to be the optimal auditory filter in terms of minimal uncertainty if the time-scale representation is calculated in the auditory system. The gammatone function is the first-order approximation of the gammachirp function. The gammachirp function has an asymmetrical Fourier power spectrum and fits the psychoacoustic auditory filter shape better than the gammatone function. Wavelet configuration is optimal for the auditory filterbank in terms of invariability in the scale representation. Data for equivalent rectangular bandwidth (ERB) suggest the wavelet gammachirp filterbank is feasible for characteristic frequencies above 800 Hz.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"425 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":"115644410","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":"Image model for computer simulation of sound wave behavior in an enclosure","authors":"E. Jan, J. Flanagan","doi":"10.1109/ASPAA.1995.482972","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482972","url":null,"abstract":"An image model using the ray tracing method has been developed to calculate the multipath in an enclosure up to a prescribed order of images. Non-uniform wall reflectivities can be included. For the special case of rectangular enclosures, the algorithm efficiently calculates the incident angle for each arriving signal and its associated reflection points. The performances of gradient microphones in an enclosure can be evaluated. It is noted that the number of images for a given order more nearly resembles an arithmetic series instead of a geometric series. The model is also extended to calculate the impulse response when obstacle planes are inside the enclosure.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"3 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":"116537427","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":"Predicting the quality of sound pickup using microphone arrays","authors":"J. G. Ryan","doi":"10.1109/ASPAA.1995.482981","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482981","url":null,"abstract":"This paper addresses the problem of predicting audio quality for speech pickup using a linear microphone array. Computer simulations using the image method are used to show that the low reverberation typical of a small room does not affect speech intelligibility. A linear array steered for spherical waves is shown to produce substantial increases in predicted listener preference. The inability of the array to maintain these increases over all source positions in the room is attributed to the limited range resolution of a delay-and-sum steered linear array.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"21 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":"123776629","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":"The Bark bilinear transform","authors":"Julius O. Smith, Jonathan S. Abel","doi":"10.1109/ASPAA.1995.482991","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482991","url":null,"abstract":"Use of a bilinear conformal map to achieve a frequency warping nearly identical to the Bark scale is described. Because the map takes the unit circle to itself, its form is that of an allpass transfer function. Since it is a first-order map, it preserves the model order of rational systems. A direct-form expression for computing the optimal allpass coefficient as a function of sampling rate is developed, and a filter design example is presented.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"20 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":"134580511","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"A comparison of head related transfer function interpolation methods","authors":"P.R. Runkle, M.A. Blommer, G. Wakefield","doi":"10.1109/ASPAA.1995.482965","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482965","url":null,"abstract":"In order to achieve realistic synthesized 3-dimensional acoustic fields over headphones, low-order approximations of head related transfer functions (HRTFs) are desirable not only because of the computational complexity reduction, but also because of the potential for allowing listeners to modify the low-order approximation parameters in order to generate interpolated HRTFs that optimize the source localization percept. By fitting the directional component of a HRTF, commonly known as the directional transfer function (DTF), it is possible to achieve low-order systems for the purpose of interpolating HRTFs even if the number of parameters required to approximate the entire HRTF is relatively large. The present paper compares the relative performance of interpolation methods for both all-zero and pole-zero modelling of the DTFs. With the goal of developing better interpolation methods by incorporating subjective preferences, active sensory tuning (AST), which has been previously shown to provide an efficient means of searching through moderate dimensional parameter spaces using subjective feedback, is proposed as a method of providing high quality interpolated DTFs within reasonable time constraints.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"1 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":"130765780","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":"Binaural modeling and auditory scene analysis","authors":"M. Bodden","doi":"10.1109/ASPAA.1995.482906","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482906","url":null,"abstract":"During the past years auditory scene analysis has become a focus of research on the human auditory system. The abilities of humans to analyze their auditory environment are so striking that they are attracting researchers from different areas. Traditionally, research on this topic was initiated and mainly performed by psychologists. But, besides the main interest to understand the underlying mechanisms, engineers are getting more and more involved. The reason for this is that possibilities to simulate auditory scene analysis offer new solutions to a variety of technical problems. Due to that, the model presented in this article, which gives a review of the work performed at the Institute of Acoustics of Bochum University in Germany, is mainly motivated by this problem-oriented thinking. In contrast to the majority of activities in the field of auditory scene analysis our approach is based on binaural peripheral processing. The intention is to understand and simulate binaural hearing. As a result of this work several models of the binaural auditory system have been developed, ranging from closely physiologically oriented models to rather signal-processing-motivated algorithms. The focus of work is on applications of these models to technical problems. These applications can be grouped into two topics: sound source localization and simulation of the cocktail-party-effect. This article describes a current model, shows its applications, and discusses its relation to auditory scene analysis.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"2 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":"127721992","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":"Use of psychological data in building ASA models","authors":"A. Bregman","doi":"10.1109/ASPAA.1995.482899","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482899","url":null,"abstract":"Many kinds of events in the environment give rise to acoustic pressure waves. Typically, the sounds are overlapped in time and in frequency. Yet, despite this mixing, our auditory systems can recover separate descriptions for each separate sound-producing event, a capacity called auditory scene analysis (ASA). The computational modelling of ASA has easier and harder parts. The easiest part is building mechanisms to extract the cues from the incoming data. It is a good deal harder to figure out how to make them interact properly to converge on a good grouping of the sense data, because this involves choosing an architecture for the system. The author propose to lay out a set of requirements that, from his perspective as a psychologist, should govern this choice of architecture.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"408 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":"116691618","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 evaluation of hearing-aid array processing","authors":"J. Kates","doi":"10.1109/ASPAA.1995.482902","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482902","url":null,"abstract":"Among the array-processing techniques that have been proposed for hearing aids are classical delay-and-sum beamforming, superdirective arrays, and adaptive arrays. In order to directly compare the effectiveness of the different processing strategies, a 10-cm long linear array was built using five uniformly-spaced omnidirectional microphones. This array was used to acquire speech and noise signals for array placements in a representative room. The processing comparison indicates that a superdirective array is an effective processing strategy, and that the performance of adaptive arrays may not justify the additional system complexity.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"36 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":"125189777","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}