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Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics最新文献

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Noise reduction based on spectral change 基于谱变化的降噪方法
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625605
T. Quatieri, R. Baxter
{"title":"Noise reduction based on spectral change","authors":"T. Quatieri, R. Baxter","doi":"10.1109/ASPAA.1997.625605","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625605","url":null,"abstract":"A noise reduction algorithm is designed for the aural enhancement of short-duration wideband signals. The signal of interest contains components possibly only a few milliseconds in duration and corrupted by a nonstationary noise background. The essence of the enhancement technique is a Wiener filter that uses a desired signal spectrum whose estimation adapts to the \"degree of stationarity\" of the measured signal. The degree of stationarity is derived from a short-time spectral derivative measurement, motivated by sensitivity of biological systems to-spectral change. Adaptive filter design tradeoffs are described, reflecting the accuracy of signal attack, background fidelity, and perceptual quality of the desired signal. Residual representations for binaural presentation are also considered.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122257502","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
Pulse tracking with a pitch tracker 带音高跟踪器的脉冲跟踪
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625623
Eric D. Scheirer
{"title":"Pulse tracking with a pitch tracker","authors":"Eric D. Scheirer","doi":"10.1109/ASPAA.1997.625623","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625623","url":null,"abstract":"A comparison of two models for processing sound is presented: the perceptually-based pitch model of Meddis and Hewitt (1991), and a vocoder model for rhythmic analysis by Scheirer. Similarities in the methods are noted, and it is demonstrated that the pitch model is also adequate for extracting the tempo of acoustic signals. The implications of this finding for perceptual models and signal processing systems are discussed.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"22 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116622807","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
A microelectronic core for a programmable digital hearing aid 可编程数字助听器的微电子核心
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625577
J. A. Hidalgo, J. C. Tejero, A. Daza, O. Oballe, A. Gago
{"title":"A microelectronic core for a programmable digital hearing aid","authors":"J. A. Hidalgo, J. C. Tejero, A. Daza, O. Oballe, A. Gago","doi":"10.1109/ASPAA.1997.625577","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625577","url":null,"abstract":"We introduce a core for a digital hearing aid that compensates the signal spoken in sensorineural impaired listeners with object of improving their intelligibility. The technique implemented is based on a digital analysis/synthesis of speech: we divided the input signal into short time blocks then we make a multiband analysis, non-linear amplification and synthesis based in a sinusoidal model of the voice, according to the subject's dynamic range in each band. The system works in real time and has been implemented with only one ASIC in 1/spl mu/ ES2 technology including 3 RAM memories with a capacity of 2432 bits and one 16/spl times/16 multiplier. The size of the die is 30.59 mm/sup 2/.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"310 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115915727","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 linear predictive method using extrapolated samples for modelling of voiced speech 一种利用外推样本进行语音建模的线性预测方法
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625592
S. Varho, P. Alku
{"title":"A linear predictive method using extrapolated samples for modelling of voiced speech","authors":"S. Varho, P. Alku","doi":"10.1109/ASPAA.1997.625592","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625592","url":null,"abstract":"A new method, linear prediction with extrapolated samples (LPES), is proposed for spectral estimation of voiced speech. In LPES, the nth sample of signal x(n) is predicted using its p+1 preceding samples by forming p lines of each two consecutive samples from those p+1 preceding samples. The values of these p lines are extrapolated at time instant n and treated as original p+1 preceding samples in conventional linear prediction (LP). The square of the prediction error is minimised using the autocorrelation criterion. LPES yields an all-pole filter whose order is one larger than the number of unknowns in the normal equations. This results to more accurate spectral estimation for voiced speech, especially at higher frequencies.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"35 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130534371","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
Atomic decompositions of audio signals 音频信号的原子分解
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625601
Michael Goodwin, Martin Vetterli
{"title":"Atomic decompositions of audio signals","authors":"Michael Goodwin, Martin Vetterli","doi":"10.1109/ASPAA.1997.625601","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625601","url":null,"abstract":"Signal modeling techniques ranging from basis expansions to parametric approaches have been applied to audio signal processing. Motivated by the fundamental limitations of basis expansions for representing arbitrary signal features and providing means for signal modifications, we consider decompositions in terms of functions that are both signal-adaptive and parametric in nature. Granular synthesis and sinusoidal modeling can be viewed in this light; we interpret these approaches as signal-adaptive expansions in terms of time-frequency atoms that are highly correlated to the fundamental signal structures. This leads naturally to a discussion of the matching pursuit algorithm for deriving decompositions using over complete dictionaries of time-frequency atoms; specifically, we compare expansions using Gabor atoms and damped sinusoids. Such decompositions identify important signal features and provide parametric representations that are useful for signal coding and analysis-modification-synthesis.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"20 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127563101","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: 18
Mixed nearfield/farfield beamforming: a new technique for speech acquisition in a reverberant environment 混合近场/远场波束形成:混响环境下语音采集的新技术
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625631
D. Ward, Gary W Elk
{"title":"Mixed nearfield/farfield beamforming: a new technique for speech acquisition in a reverberant environment","authors":"D. Ward, Gary W Elk","doi":"10.1109/ASPAA.1997.625631","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625631","url":null,"abstract":"In designing a microphone array for speech acquisition in a reverberant room, one is often faced with a mixed nearfield/farfield design problem, i.e., design a beamformer which can focus on a nearfield source, but which simultaneously can cancel room reverberation (which is typically modeled as isotropic farfield interference). This paper presents a new technique to solve such a problem. Using the spherical solution to the wave equation, a beamforming technique is presented to simultaneously approximate a desired nearfield beampattern and a desired farfield beampattern.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"69 10","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"113987321","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: 30
A subband noise-reduction method for enhancing speech in telephony and teleconferencing 一种用于增强电话和电话会议语音的子带降噪方法
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625590
E. Diethorn
{"title":"A subband noise-reduction method for enhancing speech in telephony and teleconferencing","authors":"E. Diethorn","doi":"10.1109/ASPAA.1997.625590","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625590","url":null,"abstract":"An algorithm is presented for reducing stationary, though possible changing, background noise in speech time series. A perfect reconstruction filter bank decomposes full-band time series into uniformly spaced subbands. A detection test is performed in each subband, and the results are used to dynamically adjust the gain applied to each subband. Upon reconstruction of the full-band time series, the background noise level is reduced relative to the speech. For many applications, 12 to 18 dB of noise reduction can be achieved in real-world settings. The method is demonstrated using examples of both telephony grade (narrowband) and teleconferencing grade (wideband) speech. As demonstrated using standard speech coders, the method can dramatically enhance the quality of coded speech.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"1131 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116715263","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: 23
Head tracked 3-D audio using loudspeakers 头部使用扬声器追踪3d音频
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625598
William G. Gardner
{"title":"Head tracked 3-D audio using loudspeakers","authors":"William G. Gardner","doi":"10.1109/ASPAA.1997.625598","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625598","url":null,"abstract":"Existing loudspeaker 3-D audio systems suffer from a fixed listening location. This paper proposes using a head tracker to steer the equalization zone to the position of the tracked listener. Sound localization experiments show that this strategy greatly improves localization when the listener is displaced from the ideal listening location, and also enables dynamic localization cues.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115415976","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: 25
Spectral envelope estimation using a penalized likelihood criterion 使用惩罚似然准则的谱包络估计
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625612
O. Cappé, M. Oudot, É. Moulines
{"title":"Spectral envelope estimation using a penalized likelihood criterion","authors":"O. Cappé, M. Oudot, É. Moulines","doi":"10.1109/ASPAA.1997.625612","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625612","url":null,"abstract":"Finding a smooth spectral envelope that connects estimated sinusoids is a topic of major importance in audio signal processing. A penalized likelihood criterion is introduced for the estimation of the spectral envelope in the presence of measurement noise. Various simulation results are presented that highlight the efficiency of the proposed performance criterion.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"759 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126941248","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
Efficient blind separation of convolved sound mixtures 卷积混合声的有效盲分离
Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics Pub Date : 1997-10-19 DOI: 10.1109/ASPAA.1997.625609
P. Smaragdis
{"title":"Efficient blind separation of convolved sound mixtures","authors":"P. Smaragdis","doi":"10.1109/ASPAA.1997.625609","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625609","url":null,"abstract":"We present an extension to recent approaches to blind source separation. Bell and Sejnowski (see Neural Computation 7, MIT Press, Cambridge, MA., 1996) proposed a robust algorithm for separating instantaneous mixtures. Extensions were proposed by Torkkola (see IEEE Workshop on Neural Networks for Signal Processing, Kyoto, Japan, 1996) and Lee et al. (See Advances in Neural Information Processing Systems 9, MIT Press, Cambridge, MA., 1997) for separating convolved mixtures but the computational overhead and the convergence behavior of these algorithms were not ideal. A frequency domain extension is presented which improves the stability and the performance of these algorithms.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"104 26","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-10-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131746417","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: 39
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