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Applications of Signal Processing to Audio and Acoustics, 1995., IEEE ASSP Workshop on

Date 15-18 Oct 1995

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Displaying Results 1 - 25 of 61
  • 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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  • Predicting the quality of sound pickup using microphone arrays

    Publication Year: 1995, Page(s):161 - 164
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (300 KB)

    This paper addresses the problem of predicting audio quality for speech pickup using a linear microphone array. Computer simulations using the image method are used to show that the low reverberation typical of a small room does not affect speech intelligibility. A linear array steered for spherical waves is shown to produce substantial increases in predicted listener preference. The inability of ... View full abstract»

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  • Sub-band coding of audio using recursively indexed quantization

    Publication Year: 1995, Page(s):187 - 190
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (260 KB)

    A low-complexity audio data compression technique using sub-band coding (SBC) and a recursively indexed quantizer (RIQ) is proposed. The characteristics of the proposed system are investigated and several relevant implementation issues are described. The objective performance of the proposed system is compared to conventional coding techniques. A real time implementation of the proposed system run... View full abstract»

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  • Multi-channel blind signal separation by decorrelation

    Publication Year: 1995, Page(s):155 - 158
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (256 KB)

    The separation of independent sources from mixed observed data is a fundamental and challenging problem. In many practical situations, observations may be modelled as linear mixtures of a number of source signals, i.e. a linear multi-input multi-output system. A typical example is speech recordings made in an acoustic environment in the presence of background noise and/or competing speakers. Other... View full abstract»

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  • Adaptive minimisation of the maximum error signal in an active control system

    Publication Year: 1995, Page(s):53 - 56
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (260 KB)

    Most active noise control systems with multiple error sensors minimise the sum of the modulus squared outputs of these sensors. The purpose of this paper is to present an adaptive algorithm which minimises an alternative cost function which, in the limit, is equal to the maximum modulus squared value of all the error sensors. We investigate the physical consequences of minimise the maximum modulus... View full abstract»

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  • A new orthonormal wavelet packet decomposition for audio coding using frequency-varying modulated lapped transforms

    Publication Year: 1995, Page(s):183 - 186
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (308 KB)

    Wavelet packet decompositions based on tree structured 2-channel filter banks with conjugate quadrature filters (CQF) have found many applications in the area of audio coding. Their time-frequency tiling is the dual of the time-varying modulated lapped transforms (MLT). We present a new orthonormal wavelet packet basis, which is constructed by the frequency-varying MLT. These can be viewed as the ... 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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  • A new method of tracking talker location for microphone arrays in near field

    Publication Year: 1995, Page(s):177 - 180
    Cited by:  Papers (1)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (224 KB)

    Speech recognisers are very sensitive to degradation of the acoustical environment. In order to limit the influence of noise, reverberation and interference, several strategies can be applied. A new method for tracking location (i.e., range and bearing) of a single talker of interest for microphone arrays in the very near field (the range of the talker is less than or nearly equal to the array len... View full abstract»

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  • Limitations of handsfree acoustic echo cancellers due to nonlinear loudspeaker distortion and enclosure vibration effects

    Publication Year: 1995, Page(s):103 - 106
    Cited by:  Papers (26)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (320 KB)

    The limitations of adaptive echo cancellers (AEC) for handsfree telephony include noise, finite precision and truncation effects, undermodelling of the acoustic impulse response, vibration of the plastic enclosure, loudspeaker nonlinearities, dynamic tracking and convergence and double-talk. This paper examines the effect that the loudspeaker nonlinearity and enclosure vibration have on the steady... View full abstract»

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  • Filter morphing for audio signal processing

    Publication Year: 1995, Page(s):217 - 221
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (404 KB)

    The coefficient encoding scheme called “ARMAdillo” for a second order direct form digital filter was previously proposed. We examine the behavior of a second order digital filter, whose coefficients are encoded using the ARMAdillo scheme, when it morphs from one frame to another. We restrict our discussion to biquadratic parametric equalizers and shelving filters. Results show that usi... View full abstract»

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  • Removing preechoes from audio recordings

    Publication Year: 1995, Page(s):147 - 150
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (432 KB)

    A method is described for the attenuation of preecho in audio recordings. Preechoes are attenuated replicas of the future signal that can usually be heard in tape or phonograph recordings in silences or near-silences preceding sudden, high amplitude sounds. The method described is based on a very simple attenuation+delay preecho model. The delay is estimated by use of a modified cross-correlation ... View full abstract»

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  • Multiple beam broadband beamforming: filter design and real-time implementation

    Publication Year: 1995, Page(s):173 - 176
    Cited by:  Papers (6)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (352 KB)

    In this paper the design and implementation of a truly broadband beamformer is described. Using only a limited number of microphones a beamformer has been developed with a reasonably homogeneous beampattern over the entire speech bandwidth. The solution is obtained as the result of a filter-and-sum operation. Beams are steered towards different listening-directions and for each direction a set of ... View full abstract»

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  • Real-time musical applications using frequency domain signal processing

    Publication Year: 1995, Page(s):230 - 233
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (352 KB)

    This paper presents real-time musical applications using the IRCAM signal processing workstation which make use of FFT/IFFT-based resynthesis for timbral transformation in a musical context. An intuitive and straightforward user interface, intended for use by musicians, has been developed by the authors in the Max programming environment. Techniques for high quality time-stretching, filtering, cro... View full abstract»

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  • On the minimum-phase approximation of head-related transfer functions

    Publication Year: 1995, Page(s):84 - 87
    Cited by:  Papers (18)  |  Patents (19)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (408 KB)

    The head-related transfer function embodies the amplitude and phase transformations accompanying a sound source from a fixed position in space to the eardrum. These transformations, which primarily result from the interaction of the acoustic wave with the complex geometry of the head and pinna, are known to be the principal determinants of source location. It is especially known that the interaura... View full abstract»

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  • The recreation of a castrato voice, Farinelli's voice

    Publication Year: 1995, Page(s):242 - 245
    Cited by:  Papers (1)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (388 KB)

    We present an original research work culminating in the high quality production of thirty-nine minutes of castrato voice, by the means of sound analysis, processing and synthesis. The goal of the project is the creation of a soundtrack for a film about a famous eighteenth century castrato. Two complementary voices-a coloratura-soprano and a counter-tenor-are used to cover the entire range and comp... 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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  • Regularized estimation of cepstrum envelope from discrete frequency points

    Publication Year: 1995, Page(s):213 - 216
    Cited by:  Papers (28)
    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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  • Binaural modeling and auditory scene analysis

    Publication Year: 1995, Page(s):31 - 34
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (380 KB)

    During the past years auditory scene analysis has become a focus of research on the human auditory system. The abilities of humans to analyze their auditory environment are so striking that they are attracting researchers from different areas. Traditionally, research on this topic was initiated and mainly performed by psychologists. But, besides the main interest to understand the underlying mecha... 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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  • Robust noise modelling with application to audio restoration

    Publication Year: 1995, Page(s):143 - 146
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (396 KB)

    New methods are presented for robust modelling of noise sources drawn from heavy-tailed or impulsive distributions, such as are commonly encountered in communications systems and corrupted audio signals. The methods are formulated for linear signal models within a Bayesian framework (although likelihood-based results are easily obtained as a subset of the Bayesian methods). Solutions are generated... 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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  • A simple adaptive first-order differential microphone

    Publication Year: 1995, Page(s):169 - 172
    Cited by:  Papers (36)  |  Patents (16)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (312 KB)

    As communication devices become more portable and used in any environment, the acoustic pick-up by electroacoustic transducers will require the combination of small compact transducers and signal-processing to allow high quality communication. This paper covers the design and implementation of a novel adaptive first-order differential microphone. The self-optimization is based on minimizing the mi... 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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  • Overview of the commuted piano synthesis technique

    Publication Year: 1995, Page(s):226 - 229
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (352 KB)

    The “commuted piano synthesis” algorithm is described, based on a simplified acoustic model of the piano. The model includes multiple coupled strings, a nonlinear hammer, and a linear enclosure model (including the soundboard) which can have arbitrarily large order. Simplifications are employed which greatly reduce computational complexity. Most of the simplifications are made possible... 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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