Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

17-20 Oct. 1993

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  • Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

    Publication Year: 1993
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    Freely Available from IEEE
  • Improving joint stereo audio coding by adaptive inter-channel prediction

    Publication Year: 1993, Page(s):39 - 42
    Cited by:  Papers (8)  |  Patents (38)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (312 KB)

    A method for exploiting inter-channel redundancies of stereophonic or multichannel audio signals is presented. In contrast to known stereo redundancy reduction techniques used in joint stereo audio coding. Where only the statistical dependencies between two concurrent samples of the left and right channel signals are considered, the adaptive inter-channel prediction also takes into account possibl... View full abstract»

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  • Hierarchic models of hearing for sound separation and reconstruction

    Publication Year: 1993, Page(s):157 - 160
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (304 KB)

    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... View full abstract»

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  • Local silencing of room acoustic noise using broadband active noise control

    Publication Year: 1993, Page(s):23 - 25
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (188 KB)

    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 cont... View full abstract»

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  • Current and future standardization of high-quality digital audio coding in MPEG

    Publication Year: 1993, Page(s):43 - 46
    Cited by:  Papers (3)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (340 KB)

    Since 1988 ISO/IEC JTCI/SC29 WG11 (MPEG) is working on the standardization of video and audio signals. The Audio subgroup of MPEG is working on bit rate reduction systems for high quality digital audio. Since the first phase of this standardization effort has been finished, MPEG/Audio is extending its work to multichannel audio coding systems as well as to medium quality coding at lower sampling f... View full abstract»

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  • Constraint based audio interpolators

    Publication Year: 1993, Page(s):161 - 164
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (272 KB)

    In audio digital signal processing, interpolators are used for a variety of functions, including sample rate conversion. Linear interpolation is commonly used, but has serious signal quality problems for signals with significant high frequency content. Higher order interpolators based on sine functions or other conventional lowpass filter design techniques offer somewhat better performance, but ar... View full abstract»

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  • Frequency-independent beamforming

    Publication Year: 1993, Page(s):60 - 63
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (284 KB)

    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 ... View full abstract»

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  • Interpolation of forced structural responses from non-uniform sparse measurements

    Publication Year: 1993, Page(s):26 - 29
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (264 KB)

    This paper presents a method for interpolating a sparse set of nonuniformly spaced velocity measurements on the surface of a vibrating structure. The method utilizes knowledge of the physical nature of the vibrating structure specified in terms of a given bound on the energy of the excitation forces, estimated mobilities of the structure and a known set of sparse velocity measurements. To minimize... View full abstract»

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  • Aspects of current standardization activities for high-quality, low-rate multi-channel audio coding

    Publication Year: 1993, Page(s):47 - 50
    Cited by:  Papers (3)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (348 KB)

    This paper analyzes directions in the current standardization activities for multi-channel audio, briefly reviews the composite coding schemes AC-3 and ISO 11172-3 compatible systems, and discusses requirements, features, and time-tables for the audio systems in the ISO/Moving pictures Expert Group (MPEG) phase 2 and the United States high definition television (HDTV) standardization processes View full abstract»

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  • Computation of modulation spectra for the speech transmission index using real speech

    Publication Year: 1993, Page(s):110 - 113
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (236 KB)

    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 comput... View full abstract»

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  • Generalized overlap-add sinusoidal modeling applied to quasi-harmonic tone synthesis

    Publication Year: 1993, Page(s):165 - 168
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (288 KB)

    Analysis-by-synthesis/overlap-add (ABS/OLA) sinusoidal modeling has been successfully demonstrated as an accurate, flexible, and computationally tractable representation for the purposes of speech modification and harmonic tone synthesis; however, the model formulation used to synthesize these signals does not take full advantage of the structure of quasi-harmonic music signals. This paper describ... View full abstract»

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  • A new technique to measure electroacoustic transducer directivity indices in reverberant fields

    Publication Year: 1993, Page(s):64 - 67
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (268 KB)

    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, th... View full abstract»

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  • Autocorrelation method for high-quality time/pitch-scaling

    Publication Year: 1993, Page(s):131 - 134
    Cited by:  Papers (9)  |  Patents (18)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (356 KB)

    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 ... View full abstract»

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  • Adaptive predictive coding with transform domain quantization using block size adaptation and high-resolution spectral modeling

    Publication Year: 1993, Page(s):31 - 34
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (252 KB)

    The adaptive predictive coding with transform domain quantization (APC-TQ) technique was proposed by Bhaskar (1991) for the compression of audio signals. Since then, significant developments have taken place leading to a reduction in the coding rate. While enhancing the audio quality. These developments include (i) the use of block size adaptation to exploit the variations in the stationarity of t... View full abstract»

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  • The restoration of pitch variation defects in gramophone recordings

    Publication Year: 1993, Page(s):148 - 151
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (264 KB)

    A new algorithm is presented for the identification and restoration of time-varying pitch defects in audio signals. The problem is commonly encountered as `wow' in gramophone disc and magnetic tape recordings where motor speed variations or eccentricity in the recording process are significant. The algorithm operates in two stages, the first of which trades tonal components in musical signals to g... View full abstract»

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  • A microphone array for multimedia applications

    Publication Year: 1993, Page(s):52 - 55
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (224 KB)

    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 arra... View full abstract»

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  • A time-frequency neutral network layered model for hearing perception

    Publication Year: 1993, Page(s):123 - 126
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (260 KB)

    This paper introduces a layered neural network model for hearing perception. It is based on five important perceptual properties of hearing. The neural network model processes a joint-domain representation of the input signal to yield the desired perceptual properties. The focus is on the first two layers of the model, the transformation layer and two feature extraction layers View full abstract»

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  • Developments in transaural stereo

    Publication Year: 1993, Page(s):114 - 117
    Cited by:  Papers (26)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (240 KB)

    Transaural stereo achieves precision 3-D imaging by compensating for spectral distortions in the loudspeaker-to-car signal paths. The heart of transaural stereo, signal processing for crosstalk cancellation, is herein generalized to accommodate any number of loudspeakers and listeners in any layout. Transaural equations are written and then solved using standard algebraic methods. Worked-out examp... View full abstract»

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  • HNM: a simple, efficient harmonic+noise model for speech

    Publication Year: 1993, Page(s):169 - 172
    Cited by:  Papers (13)  |  Patents (6)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (336 KB)

    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) ... View full abstract»

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  • Parametric approximation of room impulse responses based on wavelet decomposition

    Publication Year: 1993, Page(s):68 - 71
    Cited by:  Papers (1)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (240 KB)

    A new approach to the approximation and real-time simulation of room impulse responses is presented. Based on wavelet decomposition of measured impulse response data an energy-time-frequency representation of the system room is obtained. The wavelet coefficients in the frequency subbands are calculated by a multirate analysis filter bank providing aliasing-free subband processing and linear-phase ... View full abstract»

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  • The 2-D digital waveguide mesh

    Publication Year: 1993, Page(s):177 - 180
    Cited by:  Papers (48)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (256 KB)

    An extremely efficient method for modeling wave propagation in a membrane is provided by the multidimensional extension of the digital waveguide. The 2-D digital waveguide mesh is constructed out of bi-directional delay units and scattering junctions. We show that it coincides with the standard finite difference scheme in the lossless case. Wave propagation in the mesh is compared with wave propag... View full abstract»

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  • Analog/digital hybrid VLSI signal processing using single BIT modulators

    Publication Year: 1993, Page(s):136 - 139
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (180 KB)

    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 View full abstract»

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  • Robust adaptive processing of microphone array data for hearing aids

    Publication Year: 1993, Page(s):77 - 80
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (276 KB)

    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 h... View full abstract»

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  • A fast converging, low complexity adaptive filtering algorithm

    Publication Year: 1993, Page(s):4 - 7
    Cited by:  Papers (18)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (260 KB)

    This paper introduces a new adaptive filtering algorithm called fast affine projections (FAP). Its main attributes include RLS (recursive least squares) like convergence and tracking with NLMS (normalized least mean squares) like complexity. This mix of complexity and performance is similar to the recently introduced fast Newton transversal filter (FNTF) algorithm. While FAP shares some similar pr... View full abstract»

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  • Hearing aids for profoundly deaf people based on a new parametric concept

    Publication Year: 1993, Page(s):89 - 92
    Cited by:  Papers (1)  |  Patents (5)
    Request permission for commercial reuse | Click to expandAbstract |PDF file iconPDF (252 KB)

    People with severe hearing loss only have a minor part of the frequency range available for reception of information in speech signals. These people do not benefit from normal hearing aids as the information in high frequency parts of the speech is not available. To overcome this problem the authors have developed a new method enabling to present information from the frequency range of interest in... View full abstract»

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