Pub Date : 1993-10-17DOI: 10.1109/ASPAA.1993.379979
Jean Laroche
A new method is described for high-quality time or pitch modifications of audio signals. The method is a simple but efficient improvement of the splice method. Thanks to its simplicity, the algorithm can be implemented to run in real-time on standard microprocessors. Informal listening tests have demonstrated the method's capability to modify high-quality audio signals without introducing audible artifacts for moderate modification factors (up to 15%).<>
{"title":"Autocorrelation method for high-quality time/pitch-scaling","authors":"Jean Laroche","doi":"10.1109/ASPAA.1993.379979","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379979","url":null,"abstract":"A new method is described for high-quality time or pitch modifications of audio signals. The method is a simple but efficient improvement of the splice method. Thanks to its simplicity, the algorithm can be implemented to run in real-time on standard microprocessors. Informal listening tests have demonstrated the method's capability to modify high-quality audio signals without introducing audible artifacts for moderate modification factors (up to 15%).<<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":"121882156","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.1993.379984
K. Payton, R. Uchanski, L. Braida
While it has been suggested that the speech transmission index (STI) for on environment may be calculated using speech rather than test signals, computational artifacts distort the speech analyses whereas they have minimal impact on analyses with test signals. This report documents some of the difficulties encountered when using speech as the probe stimulus and proposes modifications in STI computations to circumvent some of the problems.<>
{"title":"Computation of modulation spectra for the speech transmission index using real speech","authors":"K. Payton, R. Uchanski, L. Braida","doi":"10.1109/ASPAA.1993.379984","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379984","url":null,"abstract":"While it has been suggested that the speech transmission index (STI) for on environment may be calculated using speech rather than test signals, computational artifacts distort the speech analyses whereas they have minimal impact on analyses with test signals. This report documents some of the difficulties encountered when using speech as the probe stimulus and proposes modifications in STI computations to circumvent some of the problems.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"17 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":"116724172","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.1993.379985
H. Lehnert
Virtual auditory environments refer to a procedure in which auditory environments are created by means of a computer model (Lehnert & Blauert 1991). These artificial environments are perceived as being natural and they create the impression of being present in another physical space. The sense of tele-presence can greatly be improved by making these environments interactive, that is, the subject is allowed to move and act naturally in such an environment. To this end the behaviour of the subject, namely the rotations and translations of the head, has to be monitored and reacted on by the computer model. In the course of the European ESPRIT research project SCATIS (Spatially Coordinated Auditory/Tactile Interactive Scenario) a system, the SCAT-LAB, is under development which generates interactive virtual environments for the tactile and the auditory modality. In this paper some of the design aspects for the development of the auditory part of the SCAT-LAB are presented. Although being taken from a specific project, most of the results are regarded as being quite general for the task of creating interactive virtual environments with today's technology.<>
{"title":"Real-time generation of interactive virtual auditory environments","authors":"H. Lehnert","doi":"10.1109/ASPAA.1993.379985","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379985","url":null,"abstract":"Virtual auditory environments refer to a procedure in which auditory environments are created by means of a computer model (Lehnert & Blauert 1991). These artificial environments are perceived as being natural and they create the impression of being present in another physical space. The sense of tele-presence can greatly be improved by making these environments interactive, that is, the subject is allowed to move and act naturally in such an environment. To this end the behaviour of the subject, namely the rotations and translations of the head, has to be monitored and reacted on by the computer model. In the course of the European ESPRIT research project SCATIS (Spatially Coordinated Auditory/Tactile Interactive Scenario) a system, the SCAT-LAB, is under development which generates interactive virtual environments for the tactile and the auditory modality. In this paper some of the design aspects for the development of the auditory part of the SCAT-LAB are presented. Although being taken from a specific project, most of the results are regarded as being quite general for the task of creating interactive virtual environments with today's technology.<<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":"129068966","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.1993.379996
M. Goodwin
The beamwidth of a linear array decreases as frequency increases. For broadband beamformers such as microphone arrays for teleconferencing, this frequency dependence implies that signals incident on the outer portions of the main beam are subject to an undesirable lowpass filtering process. In the paper several ways of attaining beamwidth constancy are discussed, including a novel method based on superimposing several marginally steered beams to form a constant beamwidth multi-beam. This method provides an analytically tractable framework for designing realizable constant beamwidth beamformers.<>
{"title":"Frequency-independent beamforming","authors":"M. Goodwin","doi":"10.1109/ASPAA.1993.379996","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379996","url":null,"abstract":"The beamwidth of a linear array decreases as frequency increases. For broadband beamformers such as microphone arrays for teleconferencing, this frequency dependence implies that signals incident on the outer portions of the main beam are subject to an undesirable lowpass filtering process. In the paper several ways of attaining beamwidth constancy are discussed, including a novel method based on superimposing several marginally steered beams to form a constant beamwidth multi-beam. This method provides an analytically tractable framework for designing realizable constant beamwidth beamformers.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"16 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":"127027967","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.1993.379995
G. Elko
The paper presents a new method for measuring the directivity index of an electroacoustic transducer in a diffuse reverberant environment. The method that is proposed relies on the measurement of the spectral density variance of the transfer function between source and receiver. The method requires a measurement of the source/receiver transfer function, the distance between source and receiver, the directivity of either of the transducers, and an estimate of the room constant. A variant of the method eliminates the room constant variable. The modified method is a comparison technique that requires a known source and receiver directivity and the distance between source and receiver. The methods and their limitations are discussed for computer simulated rooms and actual measurements made in a reverberant room.<>
{"title":"A new technique to measure electroacoustic transducer directivity indices in reverberant fields","authors":"G. Elko","doi":"10.1109/ASPAA.1993.379995","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379995","url":null,"abstract":"The paper presents a new method for measuring the directivity index of an electroacoustic transducer in a diffuse reverberant environment. The method that is proposed relies on the measurement of the spectral density variance of the transfer function between source and receiver. The method requires a measurement of the source/receiver transfer function, the distance between source and receiver, the directivity of either of the transducers, and an estimate of the room constant. A variant of the method eliminates the room constant variable. The modified method is a comparison technique that requires a known source and receiver directivity and the distance between source and receiver. The methods and their limitations are discussed for computer simulated rooms and actual measurements made in a reverberant room.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"142 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":"123347957","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.1993.379998
Y. Mahieux, A. Gilloire, G. Le Tourneur
A microphone array for speech pick-up is presented. This array, intended to be used in multimedia workstations for hands-free communication, is based on simple acoustic principles and it has been implemented with standard technology. Performance characteristics are given in terms of acoustic behaviour; moreover, results of a listening test are presented which show that speech picked up by the array is systematically preferred to speech picked up by a standard unidirectional microphone.<>
{"title":"A microphone array for multimedia applications","authors":"Y. Mahieux, A. Gilloire, G. Le Tourneur","doi":"10.1109/ASPAA.1993.379998","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379998","url":null,"abstract":"A microphone array for speech pick-up is presented. This array, intended to be used in multimedia workstations for hands-free communication, is based on simple acoustic principles and it has been implemented with standard technology. Performance characteristics are given in terms of acoustic behaviour; moreover, results of a listening test are presented which show that speech picked up by the array is systematically preferred to speech picked up by a standard unidirectional microphone.<<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":"133963287","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.1993.379980
T. Quatieri, R. B. Dunn, T. E. Hanna
A new approach is introduced for time-scale modification of short-duration complex acoustic signals to improve their audibility. The method preserves the time-scaled temporal envelope of a signal and for enhancement capitalizes on the perceptual importance of a signal's temporal structure. The basis for the approach is a sub-band representation whose channel phases are controlled to shape the the temporal envelope of the time-scaled signal. The phase control is derived from locations of events which occur within filterbank outputs. A frame-based generalization of the method imposes phase consistency across consecutive synthesis frames. The approach is applied to synthetic and actual short-duration acoustic signals consisting of closely-spaced and overlapping sequential time components.<>
{"title":"Time-scale modification with temporal envelope invariance","authors":"T. Quatieri, R. B. Dunn, T. E. Hanna","doi":"10.1109/ASPAA.1993.379980","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379980","url":null,"abstract":"A new approach is introduced for time-scale modification of short-duration complex acoustic signals to improve their audibility. The method preserves the time-scaled temporal envelope of a signal and for enhancement capitalizes on the perceptual importance of a signal's temporal structure. The basis for the approach is a sub-band representation whose channel phases are controlled to shape the the temporal envelope of the time-scaled signal. The phase control is derived from locations of events which occur within filterbank outputs. A frame-based generalization of the method imposes phase consistency across consecutive synthesis frames. The approach is applied to synthetic and actual short-duration acoustic signals consisting of closely-spaced and overlapping sequential time components.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"13 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":"126373913","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.1993.379973
D. Ellis
In building a machine to detect and segregate individual components in sound mixtures, the best example to copy is the human auditory system. Several models of auditory organization implement various rules of psychoacoustic grouping. We propose in addition to model auditory inference as exhibited in the well-known 'phonemic restoration illusion' of Warren (1970). A hierarchy of abstracted features and source hypotheses similar to that of Nawab (1992) allows reconstruction of obliterated detail which can then be used to recreate an 'idealized' sound without corruption. A preliminary example of fitting a harmonic model to a noisy recording of a clarinet gives a very convincing resynthesis with the interference totally removed. However, there are many issues including the design of the representation and the control architecture still to be addressed in building a more general system.<>
{"title":"Hierarchic models of hearing for sound separation and reconstruction","authors":"D. Ellis","doi":"10.1109/ASPAA.1993.379973","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379973","url":null,"abstract":"In building a machine to detect and segregate individual components in sound mixtures, the best example to copy is the human auditory system. Several models of auditory organization implement various rules of psychoacoustic grouping. We propose in addition to model auditory inference as exhibited in the well-known 'phonemic restoration illusion' of Warren (1970). A hierarchy of abstracted features and source hypotheses similar to that of Nawab (1992) allows reconstruction of obliterated detail which can then be used to recreate an 'idealized' sound without corruption. A preliminary example of fitting a harmonic model to a noisy recording of a clarinet gives a very convincing resynthesis with the interference totally removed. However, there are many issues including the design of the representation and the control architecture still to be addressed in building a more general system.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"70 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":"124772134","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.1993.379991
M. Link, K. Buckley
The paper addresses the implementation of a real-time, robust, adaptive spatial filter used as a preprocessor for a monaural hearing aid. The goal of the ongoing study is the development of a processor that provides the user spatial selectivity and an attenuation of undesired interfering sources, while robustly controlling the response to a desired source. A four microphone, real-time, robust processor has been implemented and preliminary results are discussed in terms of improvement in SNR and an intelligibility measure.<>
{"title":"Robust real-time constrained hearing aid arrays","authors":"M. Link, K. Buckley","doi":"10.1109/ASPAA.1993.379991","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.379991","url":null,"abstract":"The paper addresses the implementation of a real-time, robust, adaptive spatial filter used as a preprocessor for a monaural hearing aid. The goal of the ongoing study is the development of a processor that provides the user spatial selectivity and an attenuation of undesired interfering sources, while robustly controlling the response to a desired source. A four microphone, real-time, robust processor has been implemented and preliminary results are discussed in terms of improvement in SNR and an intelligibility measure.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"32 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":"121543785","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.1993.380007
S. Nordebo, S. Nordholm, B. Bengtsson, I. Claesson
This paper describes an evaluation of an adaptive microphone array with respect to speech recognition performance in a car. The microphone array is compared to two conventional microphones of different types. The speech recognition device is aimed to be a part of a man/machine-interface between the driver and car information services.<>
{"title":"Noise reduction using an adaptive microphone array in a car-a speech recognition evaluation","authors":"S. Nordebo, S. Nordholm, B. Bengtsson, I. Claesson","doi":"10.1109/ASPAA.1993.380007","DOIUrl":"https://doi.org/10.1109/ASPAA.1993.380007","url":null,"abstract":"This paper describes an evaluation of an adaptive microphone array with respect to speech recognition performance in a car. The microphone array is compared to two conventional microphones of different types. The speech recognition device is aimed to be a part of a man/machine-interface between the driver and car information services.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"44 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":"124410075","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}
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