Pub Date : 1997-10-19DOI: 10.1109/ASPAA.1997.625581
A. Czyzewski, R. Królikowski, B. Kostek, H. Skarżyńśki, A. Lorens
Experiments that have been carried out by the authors so far point out that locating the essential components of the speech signal in a low frequency band can improve the speech intelligibility of some hearing impaired people. It is caused by the fact that such persons can maintain their hearing ability in this range of frequencies. The experiments were performed with the use of an electronic device operating in real-time with an algorithm for lowering the audio signals frequency. The satisfactory results of these tests encouraged them to run a research project that involves designing a new digital speech processor and elaborating an advanced algorithm for the spectral transposition of speech. The paper presents a new concept of a hearing aid employing the spectral transposition method. It presents the general scheme of the vocoder-based transposition of speech signals and gives a brief description of the new speech processor implementing this method.
{"title":"A method for spectral transposition of speech signal applicable in profound hearing loss","authors":"A. Czyzewski, R. Królikowski, B. Kostek, H. Skarżyńśki, A. Lorens","doi":"10.1109/ASPAA.1997.625581","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625581","url":null,"abstract":"Experiments that have been carried out by the authors so far point out that locating the essential components of the speech signal in a low frequency band can improve the speech intelligibility of some hearing impaired people. It is caused by the fact that such persons can maintain their hearing ability in this range of frequencies. The experiments were performed with the use of an electronic device operating in real-time with an algorithm for lowering the audio signals frequency. The satisfactory results of these tests encouraged them to run a research project that involves designing a new digital speech processor and elaborating an advanced algorithm for the spectral transposition of speech. The paper presents a new concept of a hearing aid employing the spectral transposition method. It presents the general scheme of the vocoder-based transposition of speech signals and gives a brief description of the new speech processor implementing this method.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"141 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":"133927393","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 : 1997-10-19DOI: 10.1109/ASPAA.1997.625593
Y. Haneda, Y. Kaneda
We propose a new method of modeling a room transfer function (RTF) that uses common acoustical poles and their residues. The common acoustical poles correspond to the resonance frequencies (eigenvalues) of the room, and their residues are composed of the eigenfunctions of the source and receiver positions in the room. Because the common acoustical poles do not depend on the source and receiver positions, this model expresses the RTF variations due to changes in the source and receiver positions by using residue variations. We also propose methods of interpolating and extrapolating RTFs based on the proposed common-acoustical-pole and residue model. Computer simulation demonstrated that unknown RTFs can be well estimated from known (measured) RTFs by using these methods.
{"title":"Interpolation and extrapolation of room transfer functions based on common acoustical poles and their residues","authors":"Y. Haneda, Y. Kaneda","doi":"10.1109/ASPAA.1997.625593","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625593","url":null,"abstract":"We propose a new method of modeling a room transfer function (RTF) that uses common acoustical poles and their residues. The common acoustical poles correspond to the resonance frequencies (eigenvalues) of the room, and their residues are composed of the eigenfunctions of the source and receiver positions in the room. Because the common acoustical poles do not depend on the source and receiver positions, this model expresses the RTF variations due to changes in the source and receiver positions by using residue variations. We also propose methods of interpolating and extrapolating RTFs based on the proposed common-acoustical-pole and residue model. Computer simulation demonstrated that unknown RTFs can be well estimated from known (measured) RTFs by using these methods.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"69 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":"133275482","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 : 1997-10-19DOI: 10.1109/ASPAA.1997.625632
P. Chu
In set-top videoconferencing, the complete videoconferencing system fits unobtrusively on top of the television. The microphone sound pickup system is one of the most important functional blocks with constraints of small size, high performance, and low cost. Persons speaking several feet away from the system must be picked up satisfactorily while noise generated internally in the system by the cooling fan and hard drive, and noise generated externally from air conditioning and nearby computers must be attenuated. In this paper, a three microphone superdirective array is described which meets these constraints. An analog highpass and lowpass filter are used to merge two of the microphone signals to form a single channel, so that a single stereo A/D converter is required to process the three microphone signals. The microphone signals are then linearly combined so as to maximize the signal-to-noise ratio, resulting in nulls steered toward nearby objectionable noise sources.
{"title":"Superdirective microphone array for a set-top videoconferencing system","authors":"P. Chu","doi":"10.1109/ASPAA.1997.625632","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625632","url":null,"abstract":"In set-top videoconferencing, the complete videoconferencing system fits unobtrusively on top of the television. The microphone sound pickup system is one of the most important functional blocks with constraints of small size, high performance, and low cost. Persons speaking several feet away from the system must be picked up satisfactorily while noise generated internally in the system by the cooling fan and hard drive, and noise generated externally from air conditioning and nearby computers must be attenuated. In this paper, a three microphone superdirective array is described which meets these constraints. An analog highpass and lowpass filter are used to merge two of the microphone signals to form a single channel, so that a single stereo A/D converter is required to process the three microphone signals. The microphone signals are then linearly combined so as to maximize the signal-to-noise ratio, resulting in nulls steered toward nearby objectionable noise sources.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"42 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":"128400644","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 : 1997-10-19DOI: 10.1109/ASPAA.1997.625618
Julius O. Smith, G. Scavone
Two "one-filter" scattering junctions are derived which provide very accurate models of woodwind toneholes in the context of a digital waveguide model. Because toneholes in the clarinet possess only one resonance and/or anti-resonance within the audio band, a second-order digital filter suffices.
{"title":"The one-filter Keefe clarinet tonehole","authors":"Julius O. Smith, G. Scavone","doi":"10.1109/ASPAA.1997.625618","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625618","url":null,"abstract":"Two \"one-filter\" scattering junctions are derived which provide very accurate models of woodwind toneholes in the context of a digital waveguide model. Because toneholes in the clarinet possess only one resonance and/or anti-resonance within the audio band, a second-order digital filter suffices.","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":"134510602","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 : 1997-10-19DOI: 10.1109/ASPAA.1997.625603
Jean Laroche, M. Dolson
The phase-vocoder is a well-known tool for the frequency domain processing of speech or audio signals, with applications such as time compression or expansion, pitch-scale modification, noise reduction, etc. In the context of time-scale or pitch-scale modification, the phase-vocoder is usually considered to yield high quality results, especially when large modification factors are used on polyphonic or non-pitched signals. However, the phase-vocoder is also known for an artifact that plagues its output, and has been described in the literature as either "phasiness", "reverberation", or "loss of presence". Research has been devoted to understanding and reducing this artifact, and solutions have been proposed which either significantly improve the quality of the output at the cost of a very high additional computation time, or are inexpensive but only marginally effective. This paper examines the problem of phasiness in the context of time-scale modification of signals, and presents two new phase synchronization schemes which are shown to both significantly improve the sound quality, and reduce the computational cost of such modifications.
{"title":"Phase-vocoder: about this phasiness business","authors":"Jean Laroche, M. Dolson","doi":"10.1109/ASPAA.1997.625603","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625603","url":null,"abstract":"The phase-vocoder is a well-known tool for the frequency domain processing of speech or audio signals, with applications such as time compression or expansion, pitch-scale modification, noise reduction, etc. In the context of time-scale or pitch-scale modification, the phase-vocoder is usually considered to yield high quality results, especially when large modification factors are used on polyphonic or non-pitched signals. However, the phase-vocoder is also known for an artifact that plagues its output, and has been described in the literature as either \"phasiness\", \"reverberation\", or \"loss of presence\". Research has been devoted to understanding and reducing this artifact, and solutions have been proposed which either significantly improve the quality of the output at the cost of a very high additional computation time, or are inexpensive but only marginally effective. This paper examines the problem of phasiness in the context of time-scale modification of signals, and presents two new phase synchronization schemes which are shown to both significantly improve the sound quality, and reduce the computational cost of such modifications.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"134 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":"133864962","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 : 1997-10-19DOI: 10.1109/ASPAA.1997.625610
C. X. Tan, H. Tachibana
A nonlinearity-tolerated neural active noise control scheme is presented. Time-space patterns are adaptively integrated within its architecture. A learning algorithm with time-delayed memory corresponding to the secondary acoustic paths is adopted. Simulation experiments with a hybrid structure of vibrating radiation and sound in an enclosure are conducted. It is demonstrated that the proposed approach can achieve effective noise attenuation over the whole spectrum of interest, even with a strong nonlinear environment, while the conventional filtered-x LMS active noise controller falls in chaos.
{"title":"Nonlinearity-tolerated active noise control using an artificial neural network","authors":"C. X. Tan, H. Tachibana","doi":"10.1109/ASPAA.1997.625610","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625610","url":null,"abstract":"A nonlinearity-tolerated neural active noise control scheme is presented. Time-space patterns are adaptively integrated within its architecture. A learning algorithm with time-delayed memory corresponding to the secondary acoustic paths is adopted. Simulation experiments with a hybrid structure of vibrating radiation and sound in an enclosure are conducted. It is demonstrated that the proposed approach can achieve effective noise attenuation over the whole spectrum of interest, even with a strong nonlinear environment, while the conventional filtered-x LMS active noise controller falls in chaos.","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":"129977054","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 : 1997-10-19DOI: 10.1109/ASPAA.1997.625579
S. Wyrsch, A. Kaelin
This paper describes a DSP implementation of a digital hearing aid realized in the frequency domain that compensates for recruitment of loudness and cancels acoustic echos. In contrast to conventional systems which are based on a noise-probe signal, our echo canceler is adapted using only the available (e.g. speech) input signal. The main problems caused by a nonlinear feedforward filter are discussed using analytical results of the steady state behavior of the closed-loop hearing-aid system. The implemented DSP system is tested with a dummy behind-the-ear (bte) hearing-aid device on a KEMAR-head and results are presented.
{"title":"A DSP implementation of a digital hearing aid with recruitment of loudness compensation and acoustic echo cancellation","authors":"S. Wyrsch, A. Kaelin","doi":"10.1109/ASPAA.1997.625579","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625579","url":null,"abstract":"This paper describes a DSP implementation of a digital hearing aid realized in the frequency domain that compensates for recruitment of loudness and cancels acoustic echos. In contrast to conventional systems which are based on a noise-probe signal, our echo canceler is adapted using only the available (e.g. speech) input signal. The main problems caused by a nonlinear feedforward filter are discussed using analytical results of the steady state behavior of the closed-loop hearing-aid system. The implemented DSP system is tested with a dummy behind-the-ear (bte) hearing-aid device on a KEMAR-head and results are presented.","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":"127624855","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 : 1997-10-19DOI: 10.1109/ASPAA.1997.625619
R. Adams
A theory of coordinated neuronal firing events is proposed that allows the low-noise transmission of analog signals through a network of coupled neurons. The inherent quantization noise of a group of neurons can be shaped in the frequency domain in such a way as to provide a high signal-to-noise ratio over some specified signal bandwidth. This shaping is accomplished by using neural interconnections that resemble lateral inhibition.
{"title":"Spectral noise-shaping in integrate-and-fire neural networks","authors":"R. Adams","doi":"10.1109/ASPAA.1997.625619","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625619","url":null,"abstract":"A theory of coordinated neuronal firing events is proposed that allows the low-noise transmission of analog signals through a network of coupled neurons. The inherent quantization noise of a group of neurons can be shaped in the frequency domain in such a way as to provide a high signal-to-noise ratio over some specified signal bandwidth. This shaping is accomplished by using neural interconnections that resemble lateral inhibition.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"77 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":"122949596","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 : 1997-10-19DOI: 10.1109/ASPAA.1997.625597
R. Duda, W. Martens
This paper examines the range dependence of the head-related transfer function (HRTF) for a simple spherical model of the head in both the time-domain and the frequency domain. The variation of low-frequency interaural level difference (ILD) with range is shown to be significant for ranges smaller than five times the sphere radius. The impulse response explains the source of the ripples in the frequency response, and provides direct evidence that the interaural time delay (ITD) is not a strong function of range. Time-delay measurements confirm the Woodworth/Schlosberg formula. Numerical analysis indicates that the HRTF is minimum phase. Thus, except for the time delay, the impulse response can be reconstructed from a simple principle components analysis of the magnitude response.
{"title":"Range-dependence of the HRTF for a spherical head","authors":"R. Duda, W. Martens","doi":"10.1109/ASPAA.1997.625597","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625597","url":null,"abstract":"This paper examines the range dependence of the head-related transfer function (HRTF) for a simple spherical model of the head in both the time-domain and the frequency domain. The variation of low-frequency interaural level difference (ILD) with range is shown to be significant for ranges smaller than five times the sphere radius. The impulse response explains the source of the ripples in the frequency response, and provides direct evidence that the interaural time delay (ITD) is not a strong function of range. Time-delay measurements confirm the Woodworth/Schlosberg formula. Numerical analysis indicates that the HRTF is minimum phase. Thus, except for the time delay, the impulse response can be reconstructed from a simple principle components analysis of the magnitude response.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"21 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":"130642589","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 : 1997-10-19DOI: 10.1109/ASPAA.1997.625588
J. Herre, J. Johnston
Historically, the choice of the optimum filterbank has been the subject of much research and discussion in the development of perceptual audio coders. Desirable properties of a good filterbank include both a good extraction of the signal's redundancy and effective utilization of that redundancy while maintaining control over perceptual demands. Often, there is a conflict between the use of perceptual constraints and the redundancy extraction, in that a filterbank with good resolution in both time and frequency is needed. Recently, a method for performing temporal noise shaping (TNS) of the error signal of a perceptual audio coder has been proposed, providing control over both the time and frequency structure of the coding noise. This paper focuses on the core part of the scheme, forming a continuously adaptive filterbank, and discusses its theoretical background, properties and limitations.
{"title":"Continuously signal-adaptive filterbank for high-quality perceptual audio coding","authors":"J. Herre, J. Johnston","doi":"10.1109/ASPAA.1997.625588","DOIUrl":"https://doi.org/10.1109/ASPAA.1997.625588","url":null,"abstract":"Historically, the choice of the optimum filterbank has been the subject of much research and discussion in the development of perceptual audio coders. Desirable properties of a good filterbank include both a good extraction of the signal's redundancy and effective utilization of that redundancy while maintaining control over perceptual demands. Often, there is a conflict between the use of perceptual constraints and the redundancy extraction, in that a filterbank with good resolution in both time and frequency is needed. Recently, a method for performing temporal noise shaping (TNS) of the error signal of a perceptual audio coder has been proposed, providing control over both the time and frequency structure of the coding noise. This paper focuses on the core part of the scheme, forming a continuously adaptive filterbank, and discusses its theoretical background, properties and limitations.","PeriodicalId":347087,"journal":{"name":"Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"19 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":"125573966","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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