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Applications of Signal Processing to Audio and Acoustics, 1999 IEEE Workshop on

Date 20-20 Oct. 1999

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Displaying Results 1 - 25 of 62
  • Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)

    Publication Year: 1999
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    Freely Available from IEEE
  • Author index

    Publication Year: 1999, Page(s): 0_14
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    Freely Available from IEEE
  • Auditory parallax effects in the HRTF for nearby sources

    Publication Year: 1999, Page(s):171 - 174
    Cited by:  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (384 KB)

    When a sound source is close, the angle of the source relative to the center of the head can differ substantially from the angle of the source relative to the ear. Since the high-frequency features of the HRTF (head related transfer function) are known to depend on angle of the source relative to the ear, this “acoustic parallax” should produce a systematic remapping of high-frequency ... View full abstract»

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  • Double-talk robust fast converging algorithms for network echo cancellation

    Publication Year: 1999, Page(s):215 - 218
    Cited by:  Patents (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (252 KB)

    Echo cancelers which cover longer impulse responses (⩾64 ms) are desirable. Long responses create a need for more rapidly converging algorithms in order to meet the specifications for network echo cancelers devised by the ITU (International Telecommunication Union). In general, faster convergence implies a higher sensitivity to near-end disturbances, especially “double-talk”. Recen... View full abstract»

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  • Interpolating head related transfer functions in the median plane

    Publication Year: 1999, Page(s):167 - 170
    Cited by:  Papers (11)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (204 KB)

    This paper describes the interpolation of head related transfer functions (HRTFs) for all direction in the median plane. The interpolation of HRTFs enables us to reduce the number of measurements for new user's HRTFs, and also reduce the data of HRTFs in auditory virtual systems. In this paper, a simple linear interpolation method and the spline interpolation method are evaluated and advantages of... View full abstract»

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  • Narrow-band interference cancellation for enhanced speaker identification

    Publication Year: 1999, Page(s):123 - 126
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (232 KB)

    While the cepstrum feature has been widely used for speaker identification (SID), studies have shown that it can be sensitive to changes in environmental conditions. Many experiments have examined the effects of additive white Gaussian noise on the cepstral feature, but few, if any, have been conducted using additive narrow-band interference. Since such interference appears in an unpredictable fas... View full abstract»

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  • Psychoacoustical excitation of the (N)LMS algorithm for acoustical system identification

    Publication Year: 1999, Page(s):211 - 214
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (160 KB)

    This paper presents an algorithm and an implementation of orthogonal perfect correlation sequences for acoustical system identification using psychoacoustical masking effects. Therefore, the common NLMS-algorithm has been modified to incorporate hidden orthogonal Ipatov- and Huffman sequences for fast system identification. Using this method, the speed and accuracy of the identification of the lou... View full abstract»

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  • Analysis and enhancement of locally harmonic signals using adaptive multi-kernel methods

    Publication Year: 1999, Page(s):163 - 166
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (292 KB)

    This paper is concerned with the analysis/synthesis and enhancement of signals that can be efficiently modeled as “quasi-harmonic”. We first provide a formal definition of quasi-harmonic signals and show that known estimation methods for such signals can be interpreted in a non-parametric local approximation framework. We then investigate the possibility of using “adaptive multi-... View full abstract»

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  • Polyphonic pitch tracking using joint Bayesian estimation of multiple frame parameters

    Publication Year: 1999, Page(s):119 - 122
    Cited by:  Papers (17)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (228 KB)

    We present a novel approach to pitch estimation and note detection in polyphonic audio signals. We pose the problem in a Bayesian probabilistic framework, which allows us to incorporate prior knowledge about the nature of musical data into the model. We exploit the high correlation between model parameters in adjacent frames of data by explicitly modelling the frequency variation over time using l... View full abstract»

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  • Studies of a wideband stereophonic acoustic echo canceler

    Publication Year: 1999, Page(s):207 - 210
    Cited by:  Papers (2)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (252 KB)

    In this paper a wideband stereophonic acoustic echo canceler is presented. The fundamental difficulty of stereophonic acoustic echo cancellation (SAEC) is described and an echo canceler based on a fast recursive least squares algorithm in a subband structure is proposed. This structure have been used in a real-time implementation, on which experiments have been performed. In the paper, simulation ... View full abstract»

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  • On the optimality of subband adaptive filters

    Publication Year: 1999, Page(s):59 - 62
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (224 KB)

    In this paper, we derive a polyphase analysis to determine the optimum filters in a subband adaptive filter (SAF) system. The structure of this optimum solution deviates from the standard SAF approach and presents its best possible solution only as an approximation. Besides this new insight into SAF error sources, the discussed analysis allows to calculate the optimum subband responses and the sta... View full abstract»

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  • Phase relationships and amplitude envelopes in auditory perception

    Publication Year: 1999, Page(s):223 - 226
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (264 KB)

    The firing rate of an inner hair cell depends on the amplitude envelope in the associated critical band. Phase relationships between clusters of sinusoids in a critical band affect this envelope. This means that sounds with identical magnitude spectra can result in different firing patterns. This may explain why a pulse train, modeled as a sum of equal amplitude cosines, sounds different than a su... View full abstract»

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  • An alternative implementation of the superdirective beamformer

    Publication Year: 1999, Page(s):7 - 10
    Cited by:  Papers (11)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (180 KB)

    We introduce a new implementation of superdirective beamformers. The new structure has the advantage of reduced computational complexity. This advantage is due to a GSC-like (generalized sidelobe canceller) scheme. Unlike the conventional GSC, the filters in the sidelobe cancelling path are fixed and can be computed in advance by using the Wiener solution. The new structure yields exactly the same... View full abstract»

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  • Bayesian restoration of quantised audio signals using a sinusoidal model with autoregressive residuals

    Publication Year: 1999, Page(s):159 - 162
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (268 KB)

    In digital audio systems, the amplitude of the signal is quantised with finite resolution. This is a nonlinear process which introduces distortion. We develop a Bayesian, model-based approach to reducing quantisation distortion when moving an audio signal to a higher resolution medium. The signal is modelled as a sum of sinusoids and an autoregressive (AR) process of unknown order. Estimation is p... View full abstract»

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  • Bayesian single channel blind deconvolution using parametric signal and channel models

    Publication Year: 1999, Page(s):151 - 154
    Cited by:  Papers (6)  |  Patents (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (236 KB)

    This paper considers single channel blind deconvolution, in which a degraded observed signal is modelled as the convolution of a non-stationary source signal with a stationary distortion operator. Recovery of the source signal from the observed signal is achieved by modelling the source signal as a time-varying autoregressive process, the distortion operator by a IIR filter, and then using a Bayes... View full abstract»

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  • SVD-based optimal filtering with applications to noise reduction in speech signals

    Publication Year: 1999, Page(s):143 - 146
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (244 KB)

    A class of SVD-based signal enhancement procedures is described, which amount to a specific optimal filtering technique for the case where the so-called `desired response' signal cannot be observed. It is shown that this optimal filter can be written as a function of the generalized singular vectors and singular values of a so-called speech and noise data matrix. A number of simple symmetry proper... View full abstract»

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  • Robust perceptual assessment of end-to-end audio quality

    Publication Year: 1999, Page(s):39 - 42
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (244 KB)

    Perceptual quality assessment models were initially developed to predict subjective quality of codecs. Experience with telephony applications has found that today's complex networks make assessment difficult. Analogue interfaces and variable delay are amongst the technologies used in current voice transmission systems-and often make the first generation of perceptual models produce inaccurate scor... View full abstract»

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  • Pitch estimation using multiple independent time-frequency windows

    Publication Year: 1999, Page(s):115 - 118
    Cited by:  Papers (10)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (292 KB)

    A system for the detection of the pitch of musical sounds at a wide pitch range and in diverse conditions is presented. The system is built upon a pitch model that calculates independent pitch estimates in separate time-frequency windows and then combines them to yield a single estimate of the pitch. Both psychoacoustic and computational experiments were carried out to determine the optimal sizes ... View full abstract»

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  • New phase-vocoder techniques for pitch-shifting, harmonizing and other exotic effects

    Publication Year: 1999, Page(s):91 - 94
    Cited by:  Papers (10)  |  Patents (29)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (272 KB)

    The phase-vocoder is usually presented as a high-quality solution for time-scale modification of signals, pitch-scale modifications usually being implemented as a combination of timescaling and sampling rate conversion. We present two new phase-vocoder-based techniques which allow direct manipulation of the signal in the frequency-domain, enabling such applications as pitch-shifting, chorusing, ha... View full abstract»

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  • On some derivations of Gibson's approach for speech enhancement

    Publication Year: 1999, Page(s):135 - 138
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (188 KB)

    This paper deals with a Kalman filter-based enhancement of a speech signal embedded in a colored noise, when using a single microphone system. Several approaches using Kalman filtering have been developed. More particularly, Gibson et al. (1991) reported an iterative method based on the so called “noise-free” state space model, which may imply the introduction of a coordinate transform... View full abstract»

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  • Broadband beamforming optimization for speech enhancement in noisy environments

    Publication Year: 1999, Page(s):19 - 22
    Cited by:  Papers (8)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (236 KB)

    We have developed a method to optimize the directional sensitivity of a filter-and-sum beamformer. The directivity of the broadband microphone array is optimized by adjusting the spatial transducer positions and the impulse response of the beamformer. We focus on the optimization of 1-dimensional arrays consisting of M omnidirectional microphones and M FIR filters each of length L. The signal sour... View full abstract»

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  • Advances in parametric audio coding

    Publication Year: 1999, Page(s):31 - 34
    Cited by:  Papers (14)  |  Patents (35)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (288 KB)

    Parametric modelling provides an efficient representation of general audio signals and is utilised in very low bit rate audio coding. It is based on the decomposition of an audio signal into components which are described by appropriate source models and represented by model parameters. Perception models are utilised in signal decomposition and model parameter coding. This paper gives a brief tuto... View full abstract»

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  • Multichannel equalization in subbands

    Publication Year: 1999, Page(s):203 - 206
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (236 KB)

    For the dereverberation of acoustic channels or the rendering of a specific sound field, the inversion of acoustics is a central problem and generally involves multichannel techniques. In this paper, we introduce a subband approach to the adaptive solution of this equalization problem. The presented method generally allows for faster convergence at lower complexity. We also address limitations of ... View full abstract»

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  • Frequency warping in the design and implementation of fixed-point audio equalizers

    Publication Year: 1999, Page(s):55 - 58
    Cited by:  Papers (5)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (236 KB)

    This paper discusses the use of frequency warping for designing and implementing a class of discrete-time filters. This technique is particularly useful for audio filters because specifications are often given on a logarithmic frequency scale. It is shown that frequency warping allows a class of recursive filters to be designed using standard FIR techniques, and naturally leads to a structure for ... View full abstract»

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  • Vocal interfaces to musical material

    Publication Year: 1999, Page(s):107 - 110
    Cited by:  Papers (12)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (212 KB)

    As the World Wide Web and the Internet becomes the dominant form of information distribution, consideration must be given to the indexing of musical material including themes, melodies, rhythm tracks and so forth. This paper describes the implementation of an algorithm for locating song titles from the vocal input of amateur singers. The prototype algorithm exceeds 90% accuracy for 9 different sin... View full abstract»

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