Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379974
H. Gierlich
Measuring procedures yielding defined and reproducible results are required to determine transfer functions for tests and registrations. On the one hand, such a test signal allowing the determination of the transfer characteristics of these systems must simulate voice properties adequately. On the other hand, such a signal must be determined exactly so that not only the transfer function in different operating modes can be measured but also the switching and switch-off times and the behaviour of such systems in duplex operation, the echo return loss and especially the temporal behaviour of echo cancelling equipment. In order to meet all these requirements, a special test signal, called the composite source signal, has been defined.<>
{"title":"Principle and application of a new test signal to determine the transfer characteristics of telecommunication systems","authors":"H. Gierlich","doi":"10.1109/ASPAA.1993.379974","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379974","url":null,"abstract":"Measuring procedures yielding defined and reproducible results are required to determine transfer functions for tests and registrations. On the one hand, such a test signal allowing the determination of the transfer characteristics of these systems must simulate voice properties adequately. On the other hand, such a signal must be determined exactly so that not only the transfer function in different operating modes can be measured but also the switching and switch-off times and the behaviour of such systems in duplex operation, the echo return loss and especially the temporal behaviour of echo cancelling equipment. In order to meet all these requirements, a special test signal, called the composite source signal, has been defined.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"20 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123382561","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}
Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379987
F. Wightman, D. Kistler
Accurate rendering of auditory objects in a virtual auditory display depends on signal processing that is based on detailed measurements of the human free-field to eardrum transfer function (HRTF). The performance of an auditory display can be severely compromised if the HRTF measurements are not made individually, for each potential user. This requirement could sharply limit the practical application of auditory display technology. Thus, we have been working to develop a standard set of HRTFs that could be used to synthesize veridical virtual auditory objects for all users. Our latest effort along those lines has involved a feature analysis of HRTFs from 15 listeners who demonstrated high proficiency localizing virtual sources. The primary objectives were to quantify the differences among HRTFs, to identify listeners with similar and different HRTFs, and to test the localizability of virtual sources synthesized from the HRTFs of an individual with closely and not closely matched HRTFs. We used a multidimensional scaling algorithm, a statistical procedure which assesses the similarity of a set of objects and/or individuals, to analyze the HRTFs of the 15 listeners. Listeners with similar HRTFs were identified and their ability to localize virtual sources synthesized from the HRTFs of a "similar" listener was evaluated. All listeners were able to localize accurately. When these same listeners were tested with virtual sources synthesized from HRTFs that were identified to be "different" by the MDS analysis. Both azimuth and elevation of virtual sources were judged less accurately. Although we were able to identify "typical" listeners from the MDS analysis, our preliminary data suggest that several alternative sets of HRTFs may be necessary to produce a usable auditory display system.<>
{"title":"Multidimensional scaling analysis of head-related transfer functions","authors":"F. Wightman, D. Kistler","doi":"10.1109/ASPAA.1993.379987","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379987","url":null,"abstract":"Accurate rendering of auditory objects in a virtual auditory display depends on signal processing that is based on detailed measurements of the human free-field to eardrum transfer function (HRTF). The performance of an auditory display can be severely compromised if the HRTF measurements are not made individually, for each potential user. This requirement could sharply limit the practical application of auditory display technology. Thus, we have been working to develop a standard set of HRTFs that could be used to synthesize veridical virtual auditory objects for all users. Our latest effort along those lines has involved a feature analysis of HRTFs from 15 listeners who demonstrated high proficiency localizing virtual sources. The primary objectives were to quantify the differences among HRTFs, to identify listeners with similar and different HRTFs, and to test the localizability of virtual sources synthesized from the HRTFs of an individual with closely and not closely matched HRTFs. We used a multidimensional scaling algorithm, a statistical procedure which assesses the similarity of a set of objects and/or individuals, to analyze the HRTFs of the 15 listeners. Listeners with similar HRTFs were identified and their ability to localize virtual sources synthesized from the HRTFs of a \"similar\" listener was evaluated. All listeners were able to localize accurately. When these same listeners were tested with virtual sources synthesized from HRTFs that were identified to be \"different\" by the MDS analysis. Both azimuth and elevation of virtual sources were judged less accurately. Although we were able to identify \"typical\" listeners from the MDS analysis, our preliminary data suggest that several alternative sets of HRTFs may be necessary to produce a usable auditory display system.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114371517","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}
Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379992
M. Hoffman
The problem of adaptively combining the outputs of an array of microphones as a single input for a hearing aid is investigated. A robust processor based on a constrained minimum variance optimization approach is used. One fundamental criteria employed in designing this robust beamformer limits the amount of cancellation of the desired signal. The results presented include the effects of acoustic headshadow, small room reverberation, microphone placement uncertainty, and desired speaker location uncertainty. Performance improvement is measured as a predicted change in the speech reception threshold (SRT) between single microphone and multi-microphone conditions. Performance improvements are demonstrated relative to the "best" single microphone in the array for block optimum and adaptive spatial filters. The performance of the block optimum arrays is shown to be attainable with adaptive implementations. A fast-attack, slow release input signal power averager allows the adaptive processor to avoid instabilities commonly experienced with nonstationary, impulsive inputs such as speech.<>
{"title":"Robust adaptive processing of microphone array data for hearing aids","authors":"M. Hoffman","doi":"10.1109/ASPAA.1993.379992","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379992","url":null,"abstract":"The problem of adaptively combining the outputs of an array of microphones as a single input for a hearing aid is investigated. A robust processor based on a constrained minimum variance optimization approach is used. One fundamental criteria employed in designing this robust beamformer limits the amount of cancellation of the desired signal. The results presented include the effects of acoustic headshadow, small room reverberation, microphone placement uncertainty, and desired speaker location uncertainty. Performance improvement is measured as a predicted change in the speech reception threshold (SRT) between single microphone and multi-microphone conditions. Performance improvements are demonstrated relative to the \"best\" single microphone in the array for block optimum and adaptive spatial filters. The performance of the block optimum arrays is shown to be attainable with adaptive implementations. A fast-attack, slow release input signal power averager allows the adaptive processor to avoid instabilities commonly experienced with nonstationary, impulsive inputs such as speech.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"73 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114839120","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}
Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.380005
Dennis R. Morgan, D. A. Quinlan
Adaptive filtering techniques are now in widespread use for a number of applications such as adaptive arrays, adaptive noise cancellation, adaptive line enhancement, adaptive modeling and system identification, adaptive equation, and adaptive echo cancellation. These techniques have also been applied to the expanding field of active noise control. In this paper, an application of active noise control to the silencing of acoustic noise in a room is considered. First, a theoretical performance analysis using measured room impulse responses is presented. Then an experiment is described that uses a commercially available active noise control signal processor.<>
{"title":"Local silencing of room acoustic noise using broadband active noise control","authors":"Dennis R. Morgan, D. A. Quinlan","doi":"10.1109/ASPAA.1993.380005","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.380005","url":null,"abstract":"Adaptive filtering techniques are now in widespread use for a number of applications such as adaptive arrays, adaptive noise cancellation, adaptive line enhancement, adaptive modeling and system identification, adaptive equation, and adaptive echo cancellation. These techniques have also been applied to the expanding field of active noise control. In this paper, an application of active noise control to the silencing of acoustic noise in a room is considered. First, a theoretical performance analysis using measured room impulse responses is presented. Then an experiment is described that uses a commercially available active noise control signal processor.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"373 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114872142","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}
Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379978
P. Dietz, L. R. Carley
A hybrid analog/digital technique for efficient VLSI implementation of signal processing systems is presented. Single bit delta sigma modulators are used to modulate analog inputs into a form which can be considered simultaneously analog and digital, and directly manipulated as such. A cross-correlator is proposed, demonstrating the compactness of VLSI signal processing systems using this approach.<>
{"title":"Analog/digital hybrid VLSI signal processing using single BIT modulators","authors":"P. Dietz, L. R. Carley","doi":"10.1109/ASPAA.1993.379978","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379978","url":null,"abstract":"A hybrid analog/digital technique for efficient VLSI implementation of signal processing systems is presented. Single bit delta sigma modulators are used to modulate analog inputs into a form which can be considered simultaneously analog and digital, and directly manipulated as such. A cross-correlator is proposed, demonstrating the compactness of VLSI signal processing systems using this approach.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"56 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123209394","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}
Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379986
E. Wenzel, S. Foster
In implementing a spatial auditory display, many engineering compromises must be made to achieve a practical system. One such compromise involves devising methods for interpolating between the head-related transfer functions (HRTFs) used to synthesize spatial stimuli in order to achieve smooth motion trajectories and locations at finer resolutions than the empirical data. The perceptual consequences of interpolation can only be assessed by psychophysical studies. This paper compares three subjects' localization judgments for stimuli synthesized from non-interpolated HRTFs. Simple linear interpolations of the empirical HRTFs, stimuli synthesized from non-interpolated minimum-phase approximations of the HRTFs, and linear interpolations of the minimum-phase HRTFs. The empirical HRTFs used were derived from a different subject (SDO) from a previous study by Wightman and Kistler (1989) and whose data are provided with the Convolvotron synthetic 3D audio system. In general, the three subjects showed the same high rates of front-back and up-down confusions that were observed in a recent experiment using non-individualized (non-interpolated) transforms from SDO. However, there were no obvious differences in localization accuracy between the different types of synthesis conditions.<>
{"title":"Perceptual consequences of interpolating head-related transfer functions during spatial synthesis","authors":"E. Wenzel, S. Foster","doi":"10.1109/ASPAA.1993.379986","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379986","url":null,"abstract":"In implementing a spatial auditory display, many engineering compromises must be made to achieve a practical system. One such compromise involves devising methods for interpolating between the head-related transfer functions (HRTFs) used to synthesize spatial stimuli in order to achieve smooth motion trajectories and locations at finer resolutions than the empirical data. The perceptual consequences of interpolation can only be assessed by psychophysical studies. This paper compares three subjects' localization judgments for stimuli synthesized from non-interpolated HRTFs. Simple linear interpolations of the empirical HRTFs, stimuli synthesized from non-interpolated minimum-phase approximations of the HRTFs, and linear interpolations of the minimum-phase HRTFs. The empirical HRTFs used were derived from a different subject (SDO) from a previous study by Wightman and Kistler (1989) and whose data are provided with the Convolvotron synthetic 3D audio system. In general, the three subjects showed the same high rates of front-back and up-down confusions that were observed in a recent experiment using non-individualized (non-interpolated) transforms from SDO. However, there were no obvious differences in localization accuracy between the different types of synthesis conditions.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"47 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125809793","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}
Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379982
P. Chitrapu, Z. Pan
We consider the problem of least squares estimation of the frequency of a single noiseless sinusoidal signal. By constraining the signal model to be an oscillatory system and derive least squares algorithm to estimate the frequency parameters. We extend the solution to the general case of multiple noiseless sinusoids and express the global solution in terms of the inverse of a Toeplitz plus Hankel matrix. We then apply the above algorithm for ultra fast estimation of the frequency of a very low frequency sine wave. Such problems arise in the digital implementations of Ring Tone detectors in automated telephony systems. In high SNR environments, we are able to obtain reasonable estimates of the frequency within a fraction of a single period of the sine wave. We derive expressions for the bias due to additive noise and also experimentally examine the effects of signal distortions.<>
{"title":"Constrained least squares estimation of sinusoidal frequencies and application to fast estimation of very low frequency tones","authors":"P. Chitrapu, Z. Pan","doi":"10.1109/ASPAA.1993.379982","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379982","url":null,"abstract":"We consider the problem of least squares estimation of the frequency of a single noiseless sinusoidal signal. By constraining the signal model to be an oscillatory system and derive least squares algorithm to estimate the frequency parameters. We extend the solution to the general case of multiple noiseless sinusoids and express the global solution in terms of the inverse of a Toeplitz plus Hankel matrix. We then apply the above algorithm for ultra fast estimation of the frequency of a very low frequency sine wave. Such problems arise in the digital implementations of Ring Tone detectors in automated telephony systems. In high SNR environments, we are able to obtain reasonable estimates of the frequency within a fraction of a single period of the sine wave. We derive expressions for the bias due to additive noise and also experimentally examine the effects of signal distortions.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"61 5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127388759","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}
Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379970
Jean Laroche, Y. Stylianou, Eric Moulines
HNM, a new analysis/modification/synthesis model based on a harmonic+noise representation of the speech signal is presented. The HNM model has several specificities: (1) HNM assumes the speech signal to be composed of a deterministic part and of a stochastic part, (2) the deterministic part is assumed to contain only harmonically related sinusoids with linearly varying complex amplitudes, and (3) the stochastic part is modeled in both the time domain and the frequency domain. The pitch-synchronous analysis technique makes use of a coarse estimate of the pitch and simultaneously calculates the various parameters of the model and refines the pitch estimate. Because the signal is decomposed into a deterministic and a stochastic part, different modification methods can be applied to each part, yielding more natural resyntheses.<>
{"title":"HNM: a simple, efficient harmonic+noise model for speech","authors":"Jean Laroche, Y. Stylianou, Eric Moulines","doi":"10.1109/ASPAA.1993.379970","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379970","url":null,"abstract":"HNM, a new analysis/modification/synthesis model based on a harmonic+noise representation of the speech signal is presented. The HNM model has several specificities: (1) HNM assumes the speech signal to be composed of a deterministic part and of a stochastic part, (2) the deterministic part is assumed to contain only harmonically related sinusoids with linearly varying complex amplitudes, and (3) the stochastic part is modeled in both the time domain and the frequency domain. The pitch-synchronous analysis technique makes use of a coarse estimate of the pitch and simultaneously calculates the various parameters of the model and refines the pitch estimate. Because the signal is decomposed into a deterministic and a stochastic part, different modification methods can be applied to each part, yielding more natural resyntheses.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129019267","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}
Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379977
R. Wannamaker, S. Lipshitz, J. Vanderkooy
It is shown that quantizing systems without feedback respond to the use of particular spectrally-shaped dither signals quite differently from those with feedback paths. For each type of system, conditions are given which ensure that the quantization error will be wide-sense stationary with no input dependence and with a predictable power spectral density function.<>
{"title":"Dithered quantizers with and without feedback","authors":"R. Wannamaker, S. Lipshitz, J. Vanderkooy","doi":"10.1109/ASPAA.1993.379977","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379977","url":null,"abstract":"It is shown that quantizing systems without feedback respond to the use of particular spectrally-shaped dither signals quite differently from those with feedback paths. For each type of system, conditions are given which ensure that the quantization error will be wide-sense stationary with no input dependence and with a predictable power spectral density function.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"8 8","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132733093","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}
Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379993
J. Kates
Microphone arrays are the most effective of the techniques that have been proposed for improving speech intelligibility in noise for the hearing impaired. Superdirective arrays are attractive since optimal performance can be obtained for a stationary random noise field. A constrained superdirective array suitable for hearing-aid applications is discussed in the paper.<>
{"title":"Superdirective arrays for hearing aids","authors":"J. Kates","doi":"10.1109/ASPAA.1993.379993","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379993","url":null,"abstract":"Microphone arrays are the most effective of the techniques that have been proposed for improving speech intelligibility in noise for the hearing impaired. Superdirective arrays are attractive since optimal performance can be obtained for a stationary random noise field. A constrained superdirective array suitable for hearing-aid applications is discussed in the paper.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"28 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129958508","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}
京公网安备 11010802042870号
京ICP备2023020795号-1
扫码关注我们
求助内容：
应助结果提醒方式：