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Applications of Signal Processing to Audio and Acoustics, 1997. 1997 IEEE ASSP Workshop on

Date 19-22 Oct. 1997

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  • Proceedings of 1997 Workshop on Applications of Signal Processing to Audio and Acoustics

    Publication Year: 1997
    Request permission for commercial reuse | PDF file iconPDF (40 KB)
    Freely Available from IEEE
  • M-band wavelet packets and filter bank trees as flexible tools in audio signal processing

    Publication Year: 1997
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (392 KB)

    This article discusses M-band wavelet packets which combine the well-known construction of 2-band wavelet packets with concepts of M-band wavelet theory. To make the resulting tilings of the time-frequency plane even more flexible, the concept of a filter bank tree (FBT) is presented. Within this framework the design of decimated filter bank cascades, realizing some arbitrary time-frequency tiling... View full abstract»

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  • Noise reduction based on spectral change

    Publication Year: 1997
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (548 KB)

    A noise reduction algorithm is designed for the aural enhancement of short-duration wideband signals. The signal of interest contains components possibly only a few milliseconds in duration and corrupted by a nonstationary noise background. The essence of the enhancement technique is a Wiener filter that uses a desired signal spectrum whose estimation adapts to the “degree of stationarity&rd... View full abstract»

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  • The continuous frequency dynamic range compressor

    Publication Year: 1997
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (344 KB)

    The typical multiband audio compressor (TMC), such as that used in many modern hearing aids, consists of a bandpass filter bank coupled to a compression circuit which applies gain to each frequency band as a function of power in that band. Generally the filter bank is designed so that the sum of magnitude responses of the filters is unity with the band edges as steep as the implementation will all... View full abstract»

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  • Noiseless coding of quantized spectral components in MPEG-2 Advanced Audio Coding

    Publication Year: 1997
    Cited by:  Papers (5)  |  Patents (73)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (360 KB)

    Advanced Audio Coding (AAC), part of ISO/MPEG-2, issued as an international standard in April, 1997. It supports single or multiple channel audio programs and delivers excellent audio quality at or below 64 kbps/channel by exploiting the compression capabilities of a high-resolution filterbank, backward-adaptive prediction, joint channel coding, nonlinear quantizers and noiseless (Huffman) coding.... View full abstract»

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  • Pulse tracking with a pitch tracker

    Publication Year: 1997
    Cited by:  Papers (4)  |  Patents (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (516 KB)

    A comparison of two models for processing sound is presented: the perceptually-based pitch model of Meddis and Hewitt (1991), and a vocoder model for rhythmic analysis by Scheirer. Similarities in the methods are noted, and it is demonstrated that the pitch model is also adequate for extracting the tempo of acoustic signals. The implications of this finding for perceptual models and signal process... View full abstract»

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  • Optimum near-field response for microphone arrays

    Publication Year: 1997
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (368 KB)

    This paper describes the effects of optimizing the weights of an arbitrary microphone array for near-field target locations. Optimum near-field weights are shown to provide increased gain for near-field sources when compared to a uniformly weighted delay-and-sum beamformer. Practical improvements in array gain due to constrained optimization are shown to be greatest at locations close to the array... View full abstract»

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  • Adaptive noise cancellation with directional microphones

    Publication Year: 1997
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (328 KB)

    The spatial correlation function between directional microphones is useful in the design and analysis of the performance of these microphones in actual acoustic noise fields. These correlation functions are well known for omnidirectional receivers, but not well known for directional receivers. This paper investigates the spatial correlation functions for Nth-order differential microphon... View full abstract»

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  • An efficient HRTF model for 3-D sound

    Publication Year: 1997
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (364 KB)

    A simple model is presented for synthesizing binaural sound from a monaural source. The model produces vertical as well as horizontal and externalization effects. The simplicity of the model permits efficient implementation, allowing for real-time multisource operation. Additionally, the parameters in the model can be adjusted to fit a particular individual's characteristics View full abstract»

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  • A linear predictive method using extrapolated samples for modelling of voiced speech

    Publication Year: 1997
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (420 KB)

    A new method, linear prediction with extrapolated samples (LPES), is proposed for spectral estimation of voiced speech. In LPES, the nth sample of signal x(n) is predicted using its p+1 preceding samples by forming p lines of each two consecutive samples from those p+1 preceding samples. The values of these p lines are extrapolated at time instant n and treated as original p+1 preceding samples in... View full abstract»

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  • Atomic decompositions of audio signals

    Publication Year: 1997
    Cited by:  Papers (3)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (552 KB)

    Signal modeling techniques ranging from basis expansions to parametric approaches have been applied to audio signal processing. Motivated by the fundamental limitations of basis expansions for representing arbitrary signal features and providing means for signal modifications, we consider decompositions in terms of functions that are both signal-adaptive and parametric in nature. Granular synthesi... View full abstract»

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  • Adaptive beamforming with partitioned frequency-domain filters

    Publication Year: 1997
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (344 KB)

    In this paper an adaptive broadband beamformer is presented which is based on a partitioned frequency-domain least-mean-square algorithm (PFDLMS). This block algorithm is known for its efficient computation and fast convergence even when the input signals are correlated. In applications where long filters are required but only a small processing delay is allowed, a frequency domain adaptive beamfo... View full abstract»

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  • Design of a broadside array for a binaural hearing aid

    Publication Year: 1997
    Cited by:  Papers (4)  |  Patents (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (412 KB)

    This paper describes the design and implementation of a binaural directional hearing aid. This hearing aid consists of a microphone array of five directional microphones integrated into the front of a pair of spectacles. The signals of the microphones are processed with the aid of double beamforming into a left-ear and a right-ear signal. The directivity pattern of the left-ear signal has its main... View full abstract»

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  • Spectral envelope estimation using a penalized likelihood criterion

    Publication Year: 1997
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (432 KB)

    Finding a smooth spectral envelope that connects estimated sinusoids is a topic of major importance in audio signal processing. A penalized likelihood criterion is introduced for the estimation of the spectral envelope in the presence of measurement noise. Various simulation results are presented that highlight the efficiency of the proposed performance criterion View full abstract»

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  • Range-dependence of the HRTF for a spherical head

    Publication Year: 1997
    Cited by:  Papers (3)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (412 KB)

    This paper examines the range dependence of the head-related transfer function (HRTF) for a simple spherical model of the head in both the time-domain and the frequency domain. The variation of low-frequency interaural level difference (ILD) with range is shown to be significant for ranges smaller than five times the sphere radius. The impulse response explains the source of the ripples in the fre... View full abstract»

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  • On the properties of temporal processing for speech in adverse environments

    Publication Year: 1997
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (452 KB)

    In this paper we report on the results that we have obtained in the application of temporal processing to speech signals. We describe what are the properties that make temporal processing an interesting and useful technique to alleviate the harmful effects that environmental factors have on speech. Though temporal processing has been used in the past, its analysis and properties have not been stud... View full abstract»

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  • Modeling binaural auditory scene analysis by a temporal fuzzy cluster analysis approach

    Publication Year: 1997
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (484 KB)

    The psychophysical based modeling approach of computational auditory scene analysis helps to understand the human auditory system and contributes to the improvement of technical acoustical systems, e.g. hearing aids and hands free telephony. In the present paper the primitive auditory scene analysis (Bregman 1990) is characterized as a cluster analysis problem. This leads to a system based on a te... View full abstract»

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  • Elimination of delay-free loops in discrete-time models of nonlinear acoustic systems

    Publication Year: 1997
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (340 KB)

    Nonlinear acoustic systems are often described by means of nonlinear maps which act as instantaneous constraints on the solutions of a system of linear differential equations. This description leads to discrete-time models exhibiting non-computable loops. This paper presents a solution to this computability problem by means of geometrical transformation of the nonlinearities and algebraic transfor... View full abstract»

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  • Removal of low frequency transient noise from old recordings using model-based signal separation techniques

    Publication Year: 1997
    Cited by:  Papers (5)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (404 KB)

    This paper is concerned with the removal of low frequency transient noise from old gramophone recordings and film sound tracks. Low frequency transients occur as a result of large breakages or discontinuities in the recorded medium which excite a long-term resonance in the playback apparatus. We present a signal separation-based approach to this problem. Audio signals and noise transients are mode... View full abstract»

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  • A pitch-based approach to time-delay estimation of reverberant speech

    Publication Year: 1997
    Cited by:  Papers (13)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (400 KB)

    Generalized cross-correlation (GCC) has been the traditional method for estimating the relative time-delay associated with speech signals received by a pair of microphones in a reverberant, noisy environment. The filtering criterion employed is either focussed on the signal degradations due to additive noise or those due exclusively to multipath channel effects. There has been relatively little su... View full abstract»

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  • Alias-free, multiresolution sinusoidal modeling for polyphonic, wideband audio

    Publication Year: 1997
    Cited by:  Papers (10)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (460 KB)

    We describe an improved method of generating more accurate sinusoidal parameters (amplitude, frequency, phase) from a wideband polyphonic audio source in a multiresolution, non-aliased fashion. This significantly improves upon previous work of sinusoidal modeling that assumes a single-pitched monophonic source, such as speech or an individual musical instrument. In addition to a more general analy... View full abstract»

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  • A microelectronic core for a programmable digital hearing aid

    Publication Year: 1997
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (540 KB)

    We introduce a core for a digital hearing aid that compensates the signal spoken in sensorineural impaired listeners with object of improving their intelligibility. The technique implemented is based on a digital analysis/synthesis of speech: we divided the input signal into short time blocks then we make a multiband analysis, non-linear amplification and synthesis based in a sinusoidal model of t... View full abstract»

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  • Fixed-point analysis and simulations of AC-3 algorithm

    Publication Year: 1997
    Cited by:  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (496 KB)

    We perform a fixed-point analysis for the Dolby AC-3 audio decoding algorithm, and determine the suitable multiplier wordlength (say, N) satisfying the required sound quality. Then, based on the similar simulations, we try to reduce the accumulator wordlength from the usual (8+2N) to (g+N+r) where g is the wordlength for overflow guard bits and r is the wordlength for rounding with the condition r... View full abstract»

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  • Mixed nearfield/farfield beamforming: a new technique for speech acquisition in a reverberant environment

    Publication Year: 1997
    Cited by:  Papers (11)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (348 KB)

    In designing a microphone array for speech acquisition in a reverberant room, one is often faced with a mixed nearfield/farfield design problem, i.e., design a beamformer which can focus on a nearfield source, but which simultaneously can cancel room reverberation (which is typically modeled as isotropic farfield interference). This paper presents a new technique to solve such a problem. Using the... View full abstract»

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  • Continuously signal-adaptive filterbank for high-quality perceptual audio coding

    Publication Year: 1997
    Cited by:  Papers (10)  |  Patents (14)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (408 KB)

    Historically, the choice of the optimum filterbank has been the subject of much research and discussion in the development of perceptual audio coders. Desirable properties of a good filterbank include both a good extraction of the signal's redundancy and effective utilization of that redundancy while maintaining control over perceptual demands. Often, there is a conflict between the use of percept... View full abstract»

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