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Proceedings of 1995 Workshop on Applications of Signal Processing to Audio and Accoustics

15-18 Oct 1995

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Displaying Results 1 - 25 of 61
  • The Bark bilinear transform

    Publication Year: 1995, Page(s):202 - 205
    Cited by:  Papers (18)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (316 KB)

    Use of a bilinear conformal map to achieve a frequency warping nearly identical to the Bark scale is described. Because the map takes the unit circle to itself, its form is that of an allpass transfer function. Since it is a first-order map, it preserves the model order of rational systems. A direct-form expression for computing the optimal allpass coefficient as a function of sampling rate is dev... View full abstract»

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  • An optimal auditory filter

    Publication Year: 1995, Page(s):198 - 201
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (360 KB)

    The optimality of the peripheral auditory filter is investigated using operator methods applied to a scale representation. A `gammachirp' function, which consists of a frequency modulated carrier and an envelope of a gamma distribution function, is found to be the optimal auditory filter in terms of minimal uncertainty if the time-scale representation is calculated in the auditory system. The gamm... View full abstract»

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  • Estimating azimuth and elevation from interaural differences

    Publication Year: 1995, Page(s):96 - 99
    Cited by:  Papers (12)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (388 KB)

    Modeling of human auditory localization has largely been limited to lateralization, or left-to-right position. This paper describes an attempt to tackle the more complicated problem of position estimation with two degrees of freedom (azimuth and elevation). Differences in interaural intensity and arrival time are extracted from the acoustic signals at the left and right eardrums, and an estimate o... View full abstract»

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  • Evaluating a model of auditory masking for applications in audio coding

    Publication Year: 1995, Page(s):195 - 197
    Cited by:  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (212 KB)

    This investigation evaluates the adequacy of an excitation pattern (EP) model of auditory-masking to account for a masking condition commonly observed in the digital coding of audio signals: namely, the masking of quantization noise by either speech or musical sounds. This condition is modeled as a case of masking of a spectrally complex target by a multi-component complex masker (either speech or... View full abstract»

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  • Simulation of room transmission functions using a triangular beam tracing computer model

    Publication Year: 1995, Page(s):253 - 256
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (260 KB)

    The triangular beam method (TBM) combines the advantages of both the mirror image source method (MISM) and the ray tracing method (RTM). In combination with a wall diffusion model, it enables one to simulate within reasonable calculation times high time-resolution reflectograms in any complex geometrical shape. The so-obtained echograms have been further processed, in order to calculate steady-sta... View full abstract»

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  • A spherical basis function neural network for pole-zero modeling of head-related transfer functions

    Publication Year: 1995, Page(s):92 - 95
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (336 KB)

    This paper describes a neural network for approximating the parameters of a pole-zero model of the head-related transfer function (HRTF). The von Mises basis function (VMBF) is described whose response depends on spherical rather than Cartesian input coordinates. The VMBF neural network is ideally suited to the problem of learning a continuous mapping from spherical coordinates to acoustic paramet... View full abstract»

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  • An evaluation of hearing-aid array processing

    Publication Year: 1995, Page(s):15 - 18
    Cited by:  Papers (3)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (320 KB)

    Among the array-processing techniques that have been proposed for hearing aids are classical delay-and-sum beamforming, superdirective arrays, and adaptive arrays. In order to directly compare the effectiveness of the different processing strategies, a 10-cm long linear array was built using five uniformly-spaced omnidirectional microphones. This array was used to acquire speech and noise signals ... View full abstract»

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  • Parallel processing of the matched-filter array for sound capture

    Publication Year: 1995, Page(s):115 - 118
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (404 KB)

    New microphone array and signal processing techniques for high quality hands-free sound capture in noisy, reverberant enclosures have been developed. The new techniques combine matched-filtering of individual sensors and parallel processing to provide acute spatial volume selectivity which is capable of mitigating the deleterious effects of noise interference and multipath distortion. The new meth... View full abstract»

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  • Auditory segregation of concurrent signals: an operational definition

    Publication Year: 1995, Page(s):39 - 42
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (324 KB)

    Auditory segregation of simultaneous, multiple (quasi-) steady-state acoustic signals which compose an ensemble implies simultaneous resolution along three “cardinal dimensions”: vectors of pitch (residue plus timbre), temporal structure, and location. It is proposed that the normalized distance between two, or several, component signals in an ensemble along any of the three dimensions... View full abstract»

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  • Optimizing the synthesis filter bank in audio coding for minimum distortion using a frequency weighted psychoacoustic criterion

    Publication Year: 1995, Page(s):191 - 194
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (380 KB)

    This paper shows the improvement brought in filter-bank based audio coding schemes by relaxing the constraints of perfect reconstruction or maximum frequency selectivity on the synthesis filter bank: here, the coding scheme is optimized according to a frequency weighted psychoacoustic criterion. This improvement is then evaluated by means of rate/distortion curves (the weighted criterion being the... View full abstract»

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  • A numerical investigation of the invertibility of room transfer functions

    Publication Year: 1995, Page(s):249 - 252
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (272 KB)

    It has been stated that the inverse filtering of room transfer functions is possible using multiple inputs under certain conditions. We address these conditions from a numerical perspective. A method is developed to determine the numerical precision of multiple input inverse filtering techniques. This method is applied to two simulated rooms, and the numerical precision is estimated as a function ... View full abstract»

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  • Regularized estimation of cepstrum envelope from discrete frequency points

    Publication Year: 1995, Page(s):213 - 216
    Cited by:  Papers (29)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (356 KB)

    This paper presents an improved method for the estimation of a continuous frequency-envelope when the value of this envelope is specified only at discrete frequencies. It is based on the Galas/Rodet (1990) approach which consists of fitting a cepstral amplitude envelope to the specified frequency points by minimizing a frequency-domain least-squares criterion. This paper introduces a regularizatio... View full abstract»

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  • Time-scale modification with inconsistent constraints

    Publication Year: 1995, Page(s):263 - 266
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (384 KB)

    A set theoretic estimation approach is introduced for time-scale modification of complex acoustic signals. The method determines a signal that meets, in a least-squared error sense, desired temporal and spectral envelope constraints that are inconsistent. These constraints are generalized within the set theoretic framework to include other signal characteristics such as instantaneous frequency and... View full abstract»

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  • Fixed filter implementation of feedback cancellation for in-the-ear hearing aids

    Publication Year: 1995, Page(s):22 - 23
    Cited by:  Papers (1)  |  Patents (9)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (132 KB)

    Acoustic feedback oscillation in hearing aids is a problem which often prevents an adequate amount of gain from being realized for some hearing aid users. A study is presented in which a feedback cancellation technique is applied to hearing aids in 13 ears. The technique consists of modeling the feedback path around the amplifier and duplicating it with an FIR (finite impulse response) filter. The... View full abstract»

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  • Two variants of the FxLMS algorithm

    Publication Year: 1995, Page(s):123 - 126
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (308 KB)

    We present a time-domain feedback analysis of the FxLMS algorithm, which has been receiving increasing attention in the literature due to its potential application in the active control of noise. In particular, we introduce a generalized FxLMS variant and derive conditions for its l 2-stability. We also show that the algorithm can in fact be regarded as a member of the class of filtered... View full abstract»

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  • Instantaneous and frequency-warped techniques for source separation and signal parametrization

    Publication Year: 1995, Page(s):47 - 50
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (280 KB)

    This paper summarizes several contributions to the study of audio signal parameterization and source separation. The philosophy of frequency-warped signal processing provides powerful means for separating the AM and FM contributions to the MS bandwidth of a complex-valued, frequency-varying sinusoid p[n], transforming it into a signal with slowly-varying parameters. The use of frequency- and harmo... View full abstract»

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  • Phase-locked vocoder

    Publication Year: 1995, Page(s):222 - 225
    Cited by:  Papers (16)  |  Patents (9)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (308 KB)

    The phase vocoder is widely used to provide high-fidelity time stretching or contraction of audio signals such as speech or monophonic musical passages. Two problems bedevil the reconstructive phase of this technique. First, the frequency estimate is usually multi-valued and one does not know how to choose which of the possible frequencies given by the analysis to use in resynthesis. Second, a sin... View full abstract»

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  • A comparison of head related transfer function interpolation methods

    Publication Year: 1995, Page(s):88 - 91
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (388 KB)

    In order to achieve realistic synthesized 3-dimensional acoustic fields over headphones, low-order approximations of head related transfer functions (HRTFs) are desirable not only because of the computational complexity reduction, but also because of the potential for allowing listeners to modify the low-order approximation parameters in order to generate interpolated HRTFs that optimize the sourc... View full abstract»

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  • The relative contribution of interaural time and magnitude cues to dynamic sound localization

    Publication Year: 1995, Page(s):80 - 83
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (472 KB)

    This paper presents preliminary data from a study examining the relative contribution of interaural time differences (ITDs) and interaural level differences (ILDs) to the localization of virtual sound sources both with and without head motion. The listeners' task was to estimate the apparent direction and distance of virtual sources presented over headphones. Stimuli were synthesized from minimum ... View full abstract»

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  • Microphone array calibration for robust adaptive processing

    Publication Year: 1995, Page(s):11 - 14
    Cited by:  Papers (1)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (352 KB)

    Significant interest exists in the use of microphone arrays for enhancing audio and speech signals. Various combinations of fixed array, adaptive array, time-domain and frequency domain processing approaches are being considered. Any well-designed sensor array processing system needs (either explicitly or implicitly) accurate modeling or measurement of the sensor responses to allow robust and effe... View full abstract»

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  • A multiresolution audio restoration algorithm

    Publication Year: 1995, Page(s):151 - 154
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (328 KB)

    This paper presents a method of restoring noisy audio signals based on a multiresolution signal representation-the multiresolution Fourier transform (MFT). This is done by using a filter that relies on detecting musically significant features in a prototype signal, which is a different performance of the same musical piece as the signal to be warped. This filter's time-scale is then warped using a... View full abstract»

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  • Perceptually optimum adaptive filter tap profiles for subband acoustic echo cancellers

    Publication Year: 1995, Page(s):111 - 114
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (348 KB)

    A method is presented for determining optimal adaptive filter tap allocation tables, or profiles, for subband acoustic echo cancellers. Given an upper bound on the number of taps in the profile, and given a desired target profile that exceeds this bound, the method yields the profile that best satisfies a constrained least-squares optimization problem. The optimization is weighted using a composit... View full abstract»

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  • A neural oscillator model of primitive auditory grouping

    Publication Year: 1995, Page(s):35 - 38
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (380 KB)

    A computational model of primitive auditory scene analysis is described, in which the grouping of peripheral frequency channels is signalled by the pattern of temporal synchronisation in a network of neural oscillators. It is demonstrated that the model is able to group acoustic components according to their fundamental frequencies and onset times. Implications for models of pitch perception are d... View full abstract»

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  • Using a fast-recursive-least-squared algorithm in a feedback-controller

    Publication Year: 1995, Page(s):61 - 64
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (256 KB)

    In active noise control applications (ANC) recursive least-squared algorithms have become more and more common. The reason is that their convergence behavior is quite independent of the statistics of the input sound signal. Using such algorithms with digital finite impulse response filters, FIR, of an order N they show a typical convergence within N samples. There exist several algorithms for FIR-... View full abstract»

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  • On the discriminability of virtual and real sound sources

    Publication Year: 1995, Page(s):76 - 79
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (272 KB)

    A method for direct comparison between virtual and real sound sources has been developed in which listeners are presented with real sources while wearing small headphones over which virtual sources may also be presented. Adequacy of this method is assessed both acoustically and psychophysically by comparing virtual free-field stimulation to actual free-field stimulation produced by loudspeakers in... View full abstract»

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