发布求助
文献互助 智能选刊 最新文献

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

筛选
英文 中文
Robust noise modelling with application to audio restoration 鲁棒噪声建模及其在音频恢复中的应用
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482977
S. Godsill, P. Rayner
{"title":"Robust noise modelling with application to audio restoration","authors":"S. Godsill, P. Rayner","doi":"10.1109/ASPAA.1995.482977","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482977","url":null,"abstract":"New methods are presented for robust modelling of noise sources drawn from heavy-tailed or impulsive distributions, such as are commonly encountered in communications systems and corrupted audio signals. The methods are formulated for linear signal models within a Bayesian framework (although likelihood-based results are easily obtained as a subset of the Bayesian methods). Solutions are generated using powerful iterative techniques. Investigations are carried out for an audio interpolation framework in which certain samples are corrupted with additive impulsive noise.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"7 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":"124328912","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
Adaptive filters based on the best matched wavelet tree 基于最佳匹配小波树的自适应滤波器
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.483005
Z. Chen, N. Erdol, F. Bao
{"title":"Adaptive filters based on the best matched wavelet tree","authors":"Z. Chen, N. Erdol, F. Bao","doi":"10.1109/ASPAA.1995.483005","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.483005","url":null,"abstract":"In applications where some prior information exists about the physical system that gives rise to the signals under consideration, simplification is to be expected in search of representational optimality. Often the purpose of processing is to separate signals from uncorrelated and correlated noise, the latter being certifiably difficult to do without the knowledge of features that enables one to distinguish the signal from noise. Under such circumstances, a priori information about the signal can be used to chose a fixed wavelet basis so that a fixed best wavelet tree can be found in which at least one scale level contains the statistical majority of the input signal energy. Such a wavelet basis is said to match the signal. It is shown that adaptive filtering using matched wavelet filter banks results in improved convergence speed as compared to adaptive filtering with no consideration of matching. The adaptation is done with the LMS algorithm. A fast algorithm is also provided to further reduce the computation complexity. Finally, an application of the method to interference and noise cancellation is presented.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"511 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":"123692078","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
Underconstrained stochastic representations for top-down computational auditory scene analysis 自顶向下计算听觉场景分析的欠约束随机表示
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482909
D. Ellis
{"title":"Underconstrained stochastic representations for top-down computational auditory scene analysis","authors":"D. Ellis","doi":"10.1109/ASPAA.1995.482909","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482909","url":null,"abstract":"I propose a structure for the first stage of a computer system capable of performing complex auditory scene analysis similar to that accomplished by human listeners. This structure contains the following innovations over previous approaches: (1) Sound is represented as discrete elements drawn from an overcomplete vocabulary encompassing both tonal and less structured sounds, designed to highlight the interdependence in the acoustic energy. (2) Through the redundancy of the basis this analysis permits and indeed requires the imposition of additional constraints, which provides for the incorporation of top-down or context-sensitive factors. (3) A modular architecture operates on an analysis-by-synthesis principle, where processes are invoked until the representation adequately accounts for the observed sound. A common goodness-of-fit criterion allows for future expansion of the system with new explanation rules, new representational elements and more abstract levels of analysis. Some initial results of applying these ideas to scenes consisting of noise bursts and dense environmental sound are presented.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"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":"125504429","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
On the discriminability of virtual and real sound sources 虚拟声源与真实声源的可辨别性
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482951
P. Zahorik, F. Wightman, D. Kistler
{"title":"On the discriminability of virtual and real sound sources","authors":"P. Zahorik, F. Wightman, D. Kistler","doi":"10.1109/ASPAA.1995.482951","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482951","url":null,"abstract":"A method for direct comparison between virtual and real sound sources has been developed in which listeners are presented with real sources while wearing small headphones over which virtual sources may also be presented. Adequacy of this method is assessed both acoustically and psychophysically by comparing virtual free-field stimulation to actual free-field stimulation produced by loudspeakers in an anechoic chamber. The effects of windowing the impulse response of the correction filter used for virtual synthesis are also examined. Using this method, it is concluded that when no windowing is applied to the correction filter impulse response (20.48 ms in duration at a 100 kHz sampling rate), stimulation resulting from virtual synthesis is indiscriminable from real sound source stimulation. As window length is decreased, successive increases in discriminability are observed, however. These psychophysical results are consistent with acoustic validation measures.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"68 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":"131969806","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: 26
Some research issues in spatial hearing 空间听力的几个研究问题
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482949
J. Loomis
{"title":"Some research issues in spatial hearing","authors":"J. Loomis","doi":"10.1109/ASPAA.1995.482949","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482949","url":null,"abstract":"Research in spatial hearing has concentrated on directional localization by observers whose heads are stationary. Developments in virtual acoustics and ecological acoustics are stimulating researchers to consider other important aspects of spatial hearing relating to the third spatial dimension (distance). This paper discusses three such issues: auditory perception of egocentric distance, auditory perception of exocentric distance, and the dependence of perceived motion on perceived distance.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"61 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":"125886664","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
Mean-square error calculations for the subband adaptive filter system 子带自适应滤波系统的均方误差计算
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482969
P. de Leon, D. Etter
{"title":"Mean-square error calculations for the subband adaptive filter system","authors":"P. de Leon, D. Etter","doi":"10.1109/ASPAA.1995.482969","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482969","url":null,"abstract":"In this paper we develop theoretical equations for calculating the reconstructed mean-square error (MSE), E[|e/sup 2/|] for the subband adaptive filter system under zero mean, unit variance white Gaussian input. These systems are seeing increased use in acoustic echo cancellation applications and by performing the calculations, we are potentially able to evaluate and compare performance for various design parameters and various modifications to the basic design.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"27 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":"121989705","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
Perceptually optimum adaptive filter tap profiles for subband acoustic echo cancellers 子带回声消除器的感知最佳自适应滤波器分锥轮廓
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482970
E. Diethorn
{"title":"Perceptually optimum adaptive filter tap profiles for subband acoustic echo cancellers","authors":"E. Diethorn","doi":"10.1109/ASPAA.1995.482970","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482970","url":null,"abstract":"A method is presented for determining optimal adaptive filter tap allocation tables, or profiles, for subband acoustic echo cancellers. Given an upper bound on the number of taps in the profile, and given a desired target profile that exceeds this bound, the method yields the profile that best satisfies a constrained least-squares optimization problem. The optimization is weighted using a composite of physical and perceptual sensitivity functions, yielding a profile that minimizes a perceptual measure of the residual echo level.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"51 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":"122356435","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: 6
A multiresolution audio restoration algorithm 一种多分辨率音频恢复算法
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482979
H.R.R. Scott, R. Wilson
{"title":"A multiresolution audio restoration algorithm","authors":"H.R.R. Scott, R. Wilson","doi":"10.1109/ASPAA.1995.482979","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482979","url":null,"abstract":"This paper presents a method of restoring noisy audio signals based on a multiresolution signal representation-the multiresolution Fourier transform (MFT). This is done by using a filter that relies on detecting musically significant features in a prototype signal, which is a different performance of the same musical piece as the signal to be warped. This filter's time-scale is then warped using a multiresolution warping algorithm to match the time-scale of the signal to be restored (the target). The warped filter is then applied to the target signal allowing the signal to pass and rejecting the noise. A method is described that allows the exploitation of the multiresolution nature of audio signals by combining restoration results from more than one scale. Results are presented showing how the warping algorithm works and the gain in SNR for the restoration of one musical piece with differing amounts of additive Gaussian noise present.","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":"130875830","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
Filter morphing for audio signal processing 用于音频信号处理的滤波器变形
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482994
Y. Ding, D. Rossum
{"title":"Filter morphing for audio signal processing","authors":"Y. Ding, D. Rossum","doi":"10.1109/ASPAA.1995.482994","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482994","url":null,"abstract":"The coefficient encoding scheme called \"ARMAdillo\" for a second order direct form digital filter was previously proposed. We examine the behavior of a second order digital filter, whose coefficients are encoded using the ARMAdillo scheme, when it morphs from one frame to another. We restrict our discussion to biquadratic parametric equalizers and shelving filters. Results show that using encoded coefficients, the morphing performance of a parametric equalizer or shelving filter is far superior to the non-encoded case.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"71 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":"125535909","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 new orthonormal wavelet packet decomposition for audio coding using frequency-varying modulated lapped transforms 一种新的基于变频调制叠加变换的音频编码正交小波包分解方法
Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics Pub Date : 1995-10-15 DOI: 10.1109/ASPAA.1995.482986
M. Purat, P. Noll
{"title":"A new orthonormal wavelet packet decomposition for audio coding using frequency-varying modulated lapped transforms","authors":"M. Purat, P. Noll","doi":"10.1109/ASPAA.1995.482986","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482986","url":null,"abstract":"Wavelet packet decompositions based on tree structured 2-channel filter banks with conjugate quadrature filters (CQF) have found many applications in the area of audio coding. Their time-frequency tiling is the dual of the time-varying modulated lapped transforms (MLT). We present a new orthonormal wavelet packet basis, which is constructed by the frequency-varying MLT. These can be viewed as the direct analogy to time-varying transforms. In contrast to the classical decomposition, the new transform shows a good bandpass behaviour without strong spectral sidelobes. Hence, it is particularly useful for the coding of audio signals. For the signals examined the coding gain is higher than that obtained with cascaded CQF filter banks. A fast algorithm for the new wavelet packet transform is possible with a polyphase implementation for the required modulated lapped transforms. This algorithm exceeds a fast CQF based wavelet packet decomposition, especially with the high frequency resolution, which is necessary in audio coding.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"74 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":"121792941","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
首页 上一页 下一页 尾页
0
×
引用
GB/T 7714-2015
复制
MLA
复制
APA
复制
导出至
BibTeX EndNote RefMan NoteFirst NoteExpress
×
提示
您的信息不完整，为了账户安全，请先补充。
现在去补充
×
提示
您因"违规操作"
具体请查看互助需知
我知道了
×
提示
现在去查看 取消
×
提示
确定
请完成安全验证×
相关产品
×
本文献相关产品
联系我们：info@booksci.cn Book学术提供免费学术资源搜索服务，方便国内外学者检索中英文文献。致力于提供最便捷和优质的服务体验。 Copyright © 2023 布克学术 All rights reserved.
京ICP备2023020795号-1
ghs 京公网安备 11010802042870号
Book学术文献互助
Book学术文献互助群
群 号：604180095
Book学术官方微信
小红书