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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 numerical investigation of the invertibility of room transfer functions 房间传递函数可逆性的数值研究
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.483001
W. Putnam, D. Rocchesso, J. Smith
{"title":"A numerical investigation of the invertibility of room transfer functions","authors":"W. Putnam, D. Rocchesso, J. Smith","doi":"10.1109/ASPAA.1995.483001","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.483001","url":null,"abstract":"It has been stated that the inverse filtering of room transfer functions is possible using multiple inputs under certain conditions. We address these conditions from a numerical perspective. A method is developed to determine the numerical precision of multiple input inverse filtering techniques. This method is applied to two simulated rooms, and the numerical precision is estimated as a function of the number of inputs and outputs into the system.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"102 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":"122455199","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
A new method of tracking talker location for microphone arrays in near field 一种近场麦克风阵列话音定位跟踪新方法
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482985
Ming Zhang, M. Er
{"title":"A new method of tracking talker location for microphone arrays in near field","authors":"Ming Zhang, M. Er","doi":"10.1109/ASPAA.1995.482985","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482985","url":null,"abstract":"Speech recognisers are very sensitive to degradation of the acoustical environment. In order to limit the influence of noise, reverberation and interference, several strategies can be applied. A new method for tracking location (i.e., range and bearing) of a single talker of interest for microphone arrays in the very near field (the range of the talker is less than or nearly equal to the array length) is proposed. In the proposed method, an array is divided into two subarrays, focusing transformations only for the talker are used, and an iteration update is proceeded. The proposed method is suitable for cases of low signal-to-noise ratio (SNR) or/and strong interfering signals (even it is stronger than the talker's signal). Computer simulations are given to support the proposed method.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"92 4 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":"132420865","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: 1
A neural oscillator model of primitive auditory grouping 原始听觉分组的神经振荡器模型
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482907
G. Brown, M. Cooke
{"title":"A neural oscillator model of primitive auditory grouping","authors":"G. Brown, M. Cooke","doi":"10.1109/ASPAA.1995.482907","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482907","url":null,"abstract":"A computational model of primitive auditory scene analysis is described, in which the grouping of peripheral frequency channels is signalled by the pattern of temporal synchronisation in a network of neural oscillators. It is demonstrated that the model is able to group acoustic components according to their fundamental frequencies and onset times. Implications for models of pitch perception are discussed.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"50 8 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":"134153306","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: 4
Separation of multiple signals in hearing aids by output decorrelation and time-delay estimation 基于输出去相关和时延估计的助听器多信号分离
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482900
P. Bamford, N. Canagarajah
{"title":"Separation of multiple signals in hearing aids by output decorrelation and time-delay estimation","authors":"P. Bamford, N. Canagarajah","doi":"10.1109/ASPAA.1995.482900","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482900","url":null,"abstract":"Sensori-neural hearing impaired listeners have difficulty in separating multiple signals or perceiving speech in background noise and hearing aids are widely used to enhance the desired signal. We present a simple two-input system to separate two signals using output decorrelation where the filter coefficients are adaptively estimated to minimise the correlation of the output signals. In particular we show that an optimal filter can be designed and the convergence behaviour can be greatly improved by using a time-delay estimation technique. The algorithm was implemented to successfully separate two signals and the results are presented.","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":"130923741","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: 1
Overview of the commuted piano synthesis technique 交换钢琴合成技术概述
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482996
Julius O. Smith, S. A. V. Duyne
{"title":"Overview of the commuted piano synthesis technique","authors":"Julius O. Smith, S. A. V. Duyne","doi":"10.1109/ASPAA.1995.482996","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482996","url":null,"abstract":"The \"commuted piano synthesis\" algorithm is described, based on a simplified acoustic model of the piano. The model includes multiple coupled strings, a nonlinear hammer, and a linear enclosure model (including the soundboard) which can have arbitrarily large order. Simplifications are employed which greatly reduce computational complexity. Most of the simplifications are made possible by the commutation of linear, time-invariant systems. However, special care must be given to the piano hammer which is highly nonlinear and therefore does not commute with other components. In its present form, a complete, two-key piano can be synthesized in real time on a single 25 MHz Motorola DSP56001 signal processing chip.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"28 4 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":"114805462","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
Phase-locked vocoder 锁相声码器
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482995
Miller S. Puckette
{"title":"Phase-locked vocoder","authors":"Miller S. Puckette","doi":"10.1109/ASPAA.1995.482995","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482995","url":null,"abstract":"The phase vocoder is widely used to provide high-fidelity time stretching or contraction of audio signals such as speech or monophonic musical passages. Two problems bedevil the reconstructive phase of this technique. First, the frequency estimate is usually multi-valued and one does not know how to choose which of the possible frequencies given by the analysis to use in resynthesis. Second, a sinusoidal component in the incoming sound always excites several channels of the analysis; the reconstructed sine waves can interfere with each other, giving rise to a reverberant sound. An improved formulation of the phase vocoder is proposed for which the first difficulty does not arise; and a means of phase-locking adjacent channels of the resynthesis is proposed which alleviates the second one.","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":"129647646","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: 108
Evaluating a model of auditory masking for applications in audio coding 评估在音频编码中应用的听觉掩蔽模型
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482989
B. Espinoza‐Varas, S.V. Cherukuri
{"title":"Evaluating a model of auditory masking for applications in audio coding","authors":"B. Espinoza‐Varas, S.V. Cherukuri","doi":"10.1109/ASPAA.1995.482989","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482989","url":null,"abstract":"This investigation evaluates the adequacy of an excitation pattern (EP) model of auditory-masking to account for a masking condition commonly observed in the digital coding of audio signals: namely, the masking of quantization noise by either speech or musical sounds. This condition is modeled as a case of masking of a spectrally complex target by a multi-component complex masker (either speech or musical sounds). The evaluation focuses on the extent to which the EP model accounts for the 10-15 dB excess masking observed when multiple maskers occur simultaneously. Excitation-pattern predictions of the masking effects of two-component maskers were compared with predictions of masking-additivity models which closely predict experimental data. The EP model underestimates the combined effects of the two maskers both when the masker frequency components fall within the signal auditory-filter bandwidth, and when they fall outside this bandwidth. A correction for masking additivity may be needed prior to applying the EP model to audio-coding situations.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"14 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":"127488110","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: 4
The tetrahedral digital waveguide mesh 四面体数字波导网格
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482998
S. Van Duyne, J.O. Smith
{"title":"The tetrahedral digital waveguide mesh","authors":"S. Van Duyne, J.O. Smith","doi":"10.1109/ASPAA.1995.482998","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482998","url":null,"abstract":"The 2D digital waveguide mesh has proven to be effective and efficient in the modeling of musical membranes and plates, particularly in combination with recent simplifications in modeling stiffness, nonlinearities, and felt mallet excitations. The rectilinear 3D extension to the mesh had been suggested, and has been applied to the case of room acoustics. However, it requires the use of 6-port scattering junctions, which make a multiply-free implementation impossible in the isotropic case. The 4-port scattering junctions of the 2D mesh required only an internal divide by 2, which could be implemented as a right shift in binary arithmetic. However, the 6-port junction requires a divide by 3. The multiply-free cases occur for N-port junctions in which N is a power of two. We propose here a tetrahedral distribution of multiply-free 4-port scattering junctions filling space much like the molecular structure of the diamond crystal, where the placement of the scattering junctions corresponds to the placement of the carbon nuclei, and the bi-directional delay units correspond to the four tetrahedrally spaced single bonds between each pair of nuclei. We show that the tetrahedral mesh is mathematically equivalent to a finite difference scheme (FDS) which approximates the 3D lossless wave equation. We further compute the frequency- and direction-dependent plane wave propagation speed dispersion error.","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":"129996596","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: 49
Simulation of room transmission functions using a triangular beam tracing computer model 用三角形光束跟踪计算机模型模拟房间传输函数
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.483002
E. De Geest, R. Garcea
{"title":"Simulation of room transmission functions using a triangular beam tracing computer model","authors":"E. De Geest, R. Garcea","doi":"10.1109/ASPAA.1995.483002","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.483002","url":null,"abstract":"The triangular beam method (TBM) combines the advantages of both the mirror image source method (MISM) and the ray tracing method (RTM). In combination with a wall diffusion model, it enables one to simulate within reasonable calculation times high time-resolution reflectograms in any complex geometrical shape. The so-obtained echograms have been further processed, in order to calculate steady-state room transmission functions. The conversion from time domain to frequency domain has not been accomplished by FFT, but by a more direct method, superimposing wavelengths on the calculated ray paths. The method is relatively fast and has been compared with boundary element (BEM) calculations, showing a very good agreement between both approaches. This direct method is especially useful for simulating room transmission functions in the medium frequency region, i.e. the transition zone between the low-frequency domain in which wave acoustics dominate and the high-frequency domain, regulated by geometrical acoustics.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"31 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":"125425091","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: 8
Real-time musical applications using frequency domain signal processing 使用频域信号处理的实时音乐应用
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482997
Zack Settel, C. Lippe
{"title":"Real-time musical applications using frequency domain signal processing","authors":"Zack Settel, C. Lippe","doi":"10.1109/ASPAA.1995.482997","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482997","url":null,"abstract":"This paper presents real-time musical applications using the IRCAM signal processing workstation which make use of FFT/IFFT-based resynthesis for timbral transformation in a musical context. An intuitive and straightforward user interface, intended for use by musicians, has been developed by the authors in the Max programming environment. Techniques for high quality time-stretching, filtering, cross-synthesis, dynamic range processing, and spectrum shaping are presented along with dynamic control structures that allow for both fine timbral modification and control of complex sound transformations using few parameters.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"13 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":"126703921","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: 11
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