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

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
  • Some properties of tail-canceling IIR filters

    Publication Year: 1997
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (348 KB)

    Infinite impulse response (IIR) recursive linear digital filters are widely used because of their low computational cost and low storage overhead requirements. Finite impulse response (FIR) filters, on the other hand, allow the possibility of implementing linear-phase linear digital filters which have constant group delay across all frequencies. The tradeoff is that to achieve similar magnitude tr... View full abstract»

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  • Perception-based bit-allocation algorithms for audio coding

    Publication Year: 1997
    Cited by:  Papers (1)  |  Patents (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (380 KB)

    We describe six algorithms for bit-allocation in audio coding. Each algorithm stems from the minimization of a different perceptually-motivated objective function. Three of these objective functions are extensions of existing ones, and three are new. Closed-form bit-allocation equations result in five cases, and an iterative approach is required in the sixth 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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  • Spectral transformation for musical tones via time domain filtering

    Publication Year: 1997
    Cited by:  Papers (1)  |  Patents (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (416 KB)

    We present a spectral transformation technique for musical tones. It can be used to modify the brightness or timbre of a musical tone. We perform spectral transformation by modifying the line spectral frequencies or LPC roots of the original spectral envelope and filtering the tone with a spectral transformation filter in the time domain. One of the application is pitch modification where frequenc... View full abstract»

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

    Publication Year: 1997
    Cited by:  Papers (6)
    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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  • Towards a model of loudness recalibration

    Publication Year: 1997
    Cited by:  Patents (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (384 KB)

    The Zwicker (1977, 1990) loudness model is a standard for predicting the loudness of a sound. This model, along with Moore and Glasberg's (see Acustica, vol.82, p.335-45, 1996) revision of it, is fairly accurate at predicting the loudness of steady-state sounds, but falls short for many types of temporally varying sounds. One temporal effect not accounted for in the Zwicker model is loudness recal... 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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  • Sound localization of concurrent and continuous speech sources in reverberant environment

    Publication Year: 1997
    Cited by:  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (444 KB)

    This paper presents a model-based method for sound localization of concurrent and continuous speech sources in a reverberant environment. A new algorithm adopted from the echo-avoidance model of the precedence effect was used to detect the echo-free onsets by specifying a generalized pattern of impulse response. Fine structure time differences were calculated from the zero-crossing points in diffe... View full abstract»

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  • Modeling the Haas effect: a first step for solving the CASA problem

    Publication Year: 1997
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (500 KB)

    Auditory scene analysis and its older cousin, the Haas/precedence effect both involve the same acoustic and auditory phenomena. In each case it is necessary to explain the ear's ability both to hear and pay attention to sources within a background of reverberations. Thus, a successful model of the Haas effect should be capable of being extended to CASA applications. We present a model based on a v... View full abstract»

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  • Atomic decompositions of audio signals

    Publication Year: 1997
    Cited by:  Papers (6)  |  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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  • 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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  • Design of fractional delay filters using convex optimization

    Publication Year: 1997
    Cited by:  Papers (10)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (384 KB)

    Fractional sample delay (FD) filters are useful and necessary in many applications, such as the accurate steering of acoustic arrays, delay lines for physical models of musical instruments, and time delay estimation. This paper addresses the design of finite impulse response (FIR) FD filters. The problem is posed as a convex optimization problem in which the maximum modulus of the complex error is... View full abstract»

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

    Publication Year: 1997
    Cited by:  Papers (13)  |  Patents (15)
    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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  • The one-filter Keefe clarinet tonehole

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

    Two “one-filter” scattering junctions are derived which provide very accurate models of woodwind toneholes in the context of a digital waveguide model. Because toneholes in the clarinet possess only one resonance and/or anti-resonance within the audio band, a second-order digital filter suffices View full abstract»

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  • A physiological ear model for specific loudness and masking

    Publication Year: 1997
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (400 KB)

    In a variety of applications the processing of arbitrary sound signals requires models for loudness perception or auditory masking with improved accuracy, compared to psychoacoustical models known so far. In general, perceptual models can only reach higher accuracy due to special assumptions concerning signal characteristics. The presented human ear model overcomes these restrictions because of th... 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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  • A DSP implementation of a digital hearing aid with recruitment of loudness compensation and acoustic echo cancellation

    Publication Year: 1997
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (460 KB)

    This paper describes a DSP implementation of a digital hearing aid realized in the frequency domain that compensates for recruitment of loudness and cancels acoustic echos. In contrast to conventional systems which are based on a noise-probe signal, our echo canceler is adapted using only the available (e.g. speech) input signal. The main problems caused by a nonlinear feedforward filter are discu... View full abstract»

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  • Virtual-loudspeakers-based multichannel sound system

    Publication Year: 1997
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (372 KB)

    We investigate the 3D virtual-loudspeakers-based multichannel sound system. This system uses the HRTFs (head related transfer functions) as the directional perception cues and makes the transmission paths transparent by using the crosstalk cancellers. We propose both the forward and feedback types of crosstalk cancellation systems and compare their complexities and performance such as equalization... 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 (14)  |  Patents (1)
    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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  • Voice source localization for automatic camera pointing system in videoconferencing

    Publication Year: 1997
    Cited by:  Papers (6)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (408 KB)

    This paper describes the voice source localization algorithm used in the PictureTel automatic camera pointing system (LimeLightTM , dynamic speech locating technology). The system uses an array of 46 cm wide and 30 cm high, which contains 4 microphones, and is mounted on top of the monitor. The three dimensional position of a sound source is calculated from the time delays of 4 pairs of... View full abstract»

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  • 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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  • Steady-state analysis of continuous adaptation systems in hearing aids

    Publication Year: 1997
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (460 KB)

    Acoustic feedback is a problem in hearing aids that contain a substantial amount of gain, hearing aids that are used in conjunction with vented or open molds, and in-the-ear hearing aids. Acoustic feedback is both annoying and reduces the maximum usable gain of hearing-aid devices. This paper studies analytically the steady-state convergence behavior of LMS-based adaptive algorithms when operating... View full abstract»

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  • Filter bank constraints for subband and frequency-domain adaptive filters

    Publication Year: 1997
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (380 KB)

    For many years now, subband and frequency-domain adaptive filtering techniques have been proposed for the cancellation of long acoustic echoes. Classical LMS based algorithms are less attractive as their computation load is higher and the convergence behaviour for coloured far-end inputs is worse. We specify 3 realization conditions for DFT modulated subband schemes. Standard subband adaptive filt... 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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