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

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Parallel processing of the matched-filter array for sound capture 用于声音捕获的匹配滤波器阵列的并行处理
E. Jan, J. Flanagan
{"title":"Parallel processing of the matched-filter array for sound capture","authors":"E. Jan, J. Flanagan","doi":"10.1109/ASPAA.1995.482971","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482971","url":null,"abstract":"New microphone array and signal processing techniques for high quality hands-free sound capture in noisy, reverberant enclosures have been developed. The new techniques combine matched-filtering of individual sensors and parallel processing to provide acute spatial volume selectivity which is capable of mitigating the deleterious effects of noise interference and multipath distortion. The new method outperforms traditional delay-and-sum beamformers which provide only directional spatial selectivity. The research additionally explores truncated matched-filtering to reduce complexity and improve sound capture quality.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"T160 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":"125664347","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: 0
The relative contribution of interaural time and magnitude cues to dynamic sound localization 声间时间和震级线索对动态声音定位的相对贡献
Elizabeth M. Wenzel
{"title":"The relative contribution of interaural time and magnitude cues to dynamic sound localization","authors":"Elizabeth M. Wenzel","doi":"10.1109/ASPAA.1995.482963","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482963","url":null,"abstract":"This paper presents preliminary data from a study examining the relative contribution of interaural time differences (ITDs) and interaural level differences (ILDs) to the localization of virtual sound sources both with and without head motion. The listeners' task was to estimate the apparent direction and distance of virtual sources presented over headphones. Stimuli were synthesized from minimum phase representations of nonindividualized directional transfer functions. During dynamic conditions, listeners were encouraged to move their heads; the position of the listener's head was tracked and the stimuli were synthesized in real time using a Convolvotron to simulate a stationary external sound source. ILDs and ITDs were either correctly or incorrectly correlated with head motion: (1) both ILDs and ITDs correctly correlated, (2) ILDs correct, ITD fixed at 0/spl deg/ azimuth and 0/spl deg/ elevation, (3) ITDs correct, ILDs fixed at 0/spl deg/, 0/spl deg/. Similar conditions were run for static conditions except that none of the cues changed with head motion. The data indicated that, compared to static conditions, head movements helped listeners to resolve confusions primarily when ILDs were correctly correlated, although a smaller effect was also seen far correct ITDs. Together with the results for static conditions, the data suggest that localization tends to be dominated by the cue that is most reliable or consistent, when reliability is defined by consistency over time as well as across frequency.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"464 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":"131557755","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
Exponentially decaying time-recursive blind deconvolution algorithm for speech dereverberation 语音去噪的指数衰减时间递归盲反卷积算法
Ping-Ya Zhao, J. Reilly
{"title":"Exponentially decaying time-recursive blind deconvolution algorithm for speech dereverberation","authors":"Ping-Ya Zhao, J. Reilly","doi":"10.1109/ASPAA.1995.482974","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482974","url":null,"abstract":"We present a time-recursive fast algorithms for adaptive blind deconvolution as applied to nonstationary speech dereverberation. The algorithm has a computational complexity of 30L+3 multiplications and divisions per recursion (MADPR) and uses an exponentially decaying data window, where L is the length of an acoustic impulse response (AIR). Besides the adaptive property, the time-recursive algorithm can also handle the so-called \"tail\" problem.","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":"132891235","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
Automated analysis and computationally efficient synthesis of acoustic guitar strings and body 自动分析和计算有效的合成原声吉他弦和身体
K. Bradley, Mu-Huo Cheng, V. Stonick
{"title":"Automated analysis and computationally efficient synthesis of acoustic guitar strings and body","authors":"K. Bradley, Mu-Huo Cheng, V. Stonick","doi":"10.1109/ASPAA.1995.482999","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482999","url":null,"abstract":"Synthesis of stringed instrument sounds that match an acoustic prototype requires models for the string oscillation and the instrument body. Cost-effective digital filters that implement physical models of oscillating strings have been extensively studied; the current challenge is in the inexpensive replication of the spectral changes induced by the body of a physical instrument. Currently reported synthesis efforts suggest that a linear body model based on sampled impulse responses of the guitar body cannot be designed to be computationally efficient. Using an advanced IIR filter design algorithm and sampled impulse responses of the guitar body, a computationally efficient model of the guitar body is determined and can be used for synthesis of an acoustic guitar sound. Digital filters model the horizontal and vertical guitar string vibrations. A sampled plucked guitar string is analyzed to extract a linear prediction error and a Fourier series representation of the string response. A segment of the linear prediction error, which captures transient plucking effects, is used as the input to the vertical string model. The Fourier series coefficients are used to synthesize one period of the steady-state response, which is used as the input to the horizontal string model. Horizontal and vertical decay rates are analyzed from the beginning and end of the sampled string data, respectively. The guitar body model is used to provide resonance and amplify the frequencies of interest.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"58 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":"115481937","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: 19
Two variants of the FxLMS algorithm FxLMS算法的两个变体
M. Rupp, A.H. Sayed
{"title":"Two variants of the FxLMS algorithm","authors":"M. Rupp, A.H. Sayed","doi":"10.1109/ASPAA.1995.482973","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482973","url":null,"abstract":"We present a time-domain feedback analysis of the FxLMS algorithm, which has been receiving increasing attention in the literature due to its potential application in the active control of noise. In particular, we introduce a generalized FxLMS variant and derive conditions for its l/sub 2/-stability. We also show that the algorithm can in fact be regarded as a member of the class of filtered-error variants. A special case of the generalized algorithm is the so-called MFxLMS recursion, which refers to a recent modification of the standard FxLMS update. While this modification significantly improves the convergence behaviour of FxLMS, it requires of the order of 3M elementary computations per time step. This is in contrast to the 2M operations required by the standard FxLMS. We suggest two new modifications that keep the computational load at the 2M level, and which present improved convergence over the FxLMS algorithm. Simulation results are included to demonstrate the points raised in the paper.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"18 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":"122348210","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
Removing preechoes from audio recordings 从录音中去除前回声
J. Laroche
{"title":"Removing preechoes from audio recordings","authors":"J. Laroche","doi":"10.1109/ASPAA.1995.482978","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482978","url":null,"abstract":"A method is described for the attenuation of preecho in audio recordings. Preechoes are attenuated replicas of the future signal that can usually be heard in tape or phonograph recordings in silences or near-silences preceding sudden, high amplitude sounds. The method described is based on a very simple attenuation+delay preecho model. The delay is estimated by use of a modified cross-correlation technique, and the value of the attenuation is obtained by a least-squares estimation. The restoration is carried out in the frequency domain by use of a nonlinear spectral subtraction technique very similar to standard noise suppression methods.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":" 6","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-10-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114051261","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
A multidimensional Householder QR algorithm for transducer placement in active noise control systems 主动噪声控制系统中换能器放置的多维户型QR算法
Larry Heck
{"title":"A multidimensional Householder QR algorithm for transducer placement in active noise control systems","authors":"Larry Heck","doi":"10.1109/ASPAA.1995.482947","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482947","url":null,"abstract":"This paper advances the state of the art in the selection of minimal configurations of sensors and actuators for active noise control systems. The method extends previous transducer selection work by (1) presenting a unified treatment of the selection and placement of large numbers of sensors and actuators in a smart structure, (2) developing computationally efficient techniques to select the best sensor-actuator pairs for multiple unknown force disturbances exciting the structure, (3) selecting the best sensors and actuators over multiple frequencies, and (4) providing bounds on the performance of the transducer selection algorithms. The approach is based on a novel multidimensional Householder QR factorization algorithm applied to the frequency response matrices that define the noise control problem. This paper presents the theoretical development of the method.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"69 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":"124947786","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
The recreation of a castrato voice, Farinelli's voice 再现阉割的声音,法里内利的声音
P. Depalle, G. García, X. Rodet
{"title":"The recreation of a castrato voice, Farinelli's voice","authors":"P. Depalle, G. García, X. Rodet","doi":"10.1109/ASPAA.1995.483000","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.483000","url":null,"abstract":"We present an original research work culminating in the high quality production of thirty-nine minutes of castrato voice, by the means of sound analysis, processing and synthesis. The goal of the project is the creation of a soundtrack for a film about a famous eighteenth century castrato. Two complementary voices-a coloratura-soprano and a counter-tenor-are used to cover the entire range and compensate for technical difficulties. Then their timbres are homogenised by using combined techniques such as phase-vocoding, additive analysis/synthesis as well as spectral envelope and pitch estimation. Thirty-nine minutes of high quality processed singing have been created for a CD of the soundtrack.","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":"129005367","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 simple adaptive first-order differential microphone 一个简单的自适应一阶差分传声器
G. Elko, Anh-Tho Nguyen Pong
{"title":"A simple adaptive first-order differential microphone","authors":"G. Elko, Anh-Tho Nguyen Pong","doi":"10.1109/ASPAA.1995.482983","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482983","url":null,"abstract":"As communication devices become more portable and used in any environment, the acoustic pick-up by electroacoustic transducers will require the combination of small compact transducers and signal-processing to allow high quality communication. This paper covers the design and implementation of a novel adaptive first-order differential microphone. The self-optimization is based on minimizing the microphone output under the constraint that the solitary null for first order systems is located in the rear-half plane. The constraint is simply realized by the judicious subtraction of time-delayed outputs from two closely-spaced omnidirectional microphones. Although the solution presented does not maximize the signal-to-noise ratio, it can significantly improve the signal-to-noise ratio in certain acoustic fields.","PeriodicalId":295140,"journal":{"name":"Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics","volume":"11 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":"116951874","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: 137
On the minimum-phase approximation of head-related transfer functions 头部相关传递函数的最小相位逼近
A. Kulkarni, S. Isabelle, H. Colburn
{"title":"On the minimum-phase approximation of head-related transfer functions","authors":"A. Kulkarni, S. Isabelle, H. Colburn","doi":"10.1109/ASPAA.1995.482964","DOIUrl":"https://doi.org/10.1109/ASPAA.1995.482964","url":null,"abstract":"The head-related transfer function embodies the amplitude and phase transformations accompanying a sound source from a fixed position in space to the eardrum. These transformations, which primarily result from the interaction of the acoustic wave with the complex geometry of the head and pinna, are known to be the principal determinants of source location. It is especially known that the interaural time differences (ITDs) are encoded by the phase spectra of the HRTFs at the two ears. In this study we explore a model of the HRTF which approximates the HRTF as a minimum phase function. The model is tested both theoretically and through psychophysical experiments. Results suggest that a minimum phase representation of the phase spectra, augmented by a position-dependent, frequency-independent ITD, is an adequate description of the HRTF phase. Both the scientific and the practical consequences of this result shall be discussed.","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":"115924886","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: 80
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