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
    Request permission for commercial reuse | PDF file iconPDF (43 KB)
    Freely Available from IEEE
  • Objective measures based on neural networks for hearing loss compensation techniques

    Publication Year: 1993, Page(s):93 - 96
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (260 KB)

    An objective measures system has been developed to predict the results of subject-based tests for sensorineural hearing loss compensation techniques. Parameters related to the loudness level of the compensated speech signal are extracted from its frequency spectrum. These parameters are then used to train a neural network based phoneme classifier. Good prediction results have been achieved for two... View full abstract»

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  • Multidimensional scaling analysis of head-related transfer functions

    Publication Year: 1993, Page(s):98 - 101
    Cited by:  Papers (4)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    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 applica... View full abstract»

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  • Perceptual consequences of interpolating head-related transfer functions during spatial synthesis

    Publication Year: 1993, Page(s):102 - 105
    Cited by:  Papers (13)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (324 KB)

    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 consequenc... View full abstract»

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  • The 2-D digital waveguide mesh

    Publication Year: 1993, Page(s):177 - 180
    Cited by:  Papers (48)  |  Patents (1)
    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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  • Real-time generation of interactive virtual auditory environments

    Publication Year: 1993, Page(s):106 - 109
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (324 KB)

    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 subje... 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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  • 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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  • 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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  • Weaver SSB subband acoustic echo canceller [videoconferencing application]

    Publication Year: 1993, Page(s):8 - 11
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (236 KB)

    A Weaver SSB subband structure is used to implement an acoustic echo canceller. The structure has 29 bands of 250 Hz width, covering the audio range from 0 to 7 kHz. The Weaver structure lowers each band pass region to baseband, allows for oversampling to eliminate aliasing components, and is computationally efficient. The subsampled components are purely real, as compared to the complex component... 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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  • 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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  • Robust real-time constrained hearing aid arrays

    Publication Year: 1993, Page(s):81 - 84
    Cited by:  Papers (1)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (292 KB)

    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 proc... 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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  • Computationally efficient compression of audio signals by means of RIQ-DPCM

    Publication Year: 1993, Page(s):35 - 38
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (256 KB)

    The need to transmit large amounts of data over limited bandwidth channels has resulted in many methods for digital data compression. The common approach is to identify and remove redundancy from the input data stream using knowledge of the source characteristics. In the case of signals intended for human observers (speech, music, pictures, etc.) it is also useful to consider the strengths and wea... 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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  • A comparison of gradient-based algorithms for echo compensation with decorrelating properties

    Publication Year: 1993, Page(s):12 - 15
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    Cancelling echoes by using the normalized least mean square (NLMS) algorithm has been state of the art for many years. In acoustical echo compensation, however, it is common to estimate more than 1000 parameters resulting in a too slow convergence when driven by speech signals. In order to overcome this drawback, a lot of modifications have been published in the last years, all having one goal: to... 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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  • Constrained least squares estimation of sinusoidal frequencies and application to fast estimation of very low frequency tones

    Publication Year: 1993, Page(s):119 - 122
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (248 KB)

    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... View full abstract»

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  • An all digital concha hearing aid

    Publication Year: 1993, Page(s):85 - 88
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (248 KB)

    The paper describes an all digital concha hearing aid. The main features of this hearing aid concept are a large vent, acoustic feed-back cancellation, great flexibility by programming, a versatile equalizer, and an advanced compressor. The A/D and D/A converters have log/in characteristics and the signal processing is performed by floating point arithmetic, ensuring a large dynamic range and a si... 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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  • Improving joint stereo audio coding by adaptive inter-channel prediction

    Publication Year: 1993, Page(s):39 - 42
    Cited by:  Papers (8)  |  Patents (39)
    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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  • 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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  • Noise reduction using an adaptive microphone array in a car-a speech recognition evaluation

    Publication Year: 1993, Page(s):16 - 18
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (180 KB)

    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 View full abstract»

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  • Directional microphones in computer simulated and real rooms

    Publication Year: 1993, Page(s):56 - 59
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (216 KB)

    The subjective effects of utilizing highly directional microphones in a teleconferencing setting are not well understood. Computer simulation of both complex microphone systems and room environments offer one opportunity to study the combined effects. A complex microphone system can be modeled as a collection of point microphones distributed in space and summed with appropriate time delays. Establ... View full abstract»

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