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Applications of Signal Processing to Audio and Acoustics, 1993. Final Program and Paper Summaries., 1993 IEEE Workshop on

Date 17-20 Oct. 1993

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Displaying Results 1 - 25 of 44
  • Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

    Publication Year: 1993
    Request permission for commercial reuse | PDF file iconPDF (43 KB)
    Freely Available from IEEE
  • Improving joint stereo audio coding by adaptive inter-channel prediction

    Publication Year: 1993, Page(s):39 - 42
    Cited by:  Papers (8)  |  Patents (38)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (312 KB)

    A method for exploiting inter-channel redundancies of stereophonic or multichannel audio signals is presented. In contrast to known stereo redundancy reduction techniques used in joint stereo audio coding. Where only the statistical dependencies between two concurrent samples of the left and right channel signals are considered, the adaptive inter-channel prediction also takes into account possibl... View full abstract»

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  • Analog/digital hybrid VLSI signal processing using single BIT modulators

    Publication Year: 1993, Page(s):136 - 139
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (180 KB)

    A hybrid analog/digital technique for efficient VLSI implementation of signal processing systems is presented. Single bit delta sigma modulators are used to modulate analog inputs into a form which can be considered simultaneously analog and digital, and directly manipulated as such. A cross-correlator is proposed, demonstrating the compactness of VLSI signal processing systems using this approach View full abstract»

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  • An all digital concha hearing aid

    Publication Year: 1993, Page(s):85 - 88
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (248 KB)

    The paper describes an all digital concha hearing aid. The main features of this hearing aid concept are a large vent, acoustic feed-back cancellation, great flexibility by programming, a versatile equalizer, and an advanced compressor. The A/D and D/A converters have log/in characteristics and the signal processing is performed by floating point arithmetic, ensuring a large dynamic range and a si... View full abstract»

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  • Robust real-time constrained hearing aid arrays

    Publication Year: 1993, Page(s):81 - 84
    Cited by:  Papers (1)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (292 KB)

    The paper addresses the implementation of a real-time, robust, adaptive spatial filter used as a preprocessor for a monaural hearing aid. The goal of the ongoing study is the development of a processor that provides the user spatial selectivity and an attenuation of undesired interfering sources, while robustly controlling the response to a desired source. A four microphone, real-time, robust proc... View full abstract»

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  • Objective measures based on neural networks for hearing loss compensation techniques

    Publication Year: 1993, Page(s):93 - 96
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (260 KB)

    An objective measures system has been developed to predict the results of subject-based tests for sensorineural hearing loss compensation techniques. Parameters related to the loudness level of the compensated speech signal are extracted from its frequency spectrum. These parameters are then used to train a neural network based phoneme classifier. Good prediction results have been achieved for two... View full abstract»

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  • Constrained least squares estimation of sinusoidal frequencies and application to fast estimation of very low frequency tones

    Publication Year: 1993, Page(s):119 - 122
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (248 KB)

    We consider the problem of least squares estimation of the frequency of a single noiseless sinusoidal signal. By constraining the signal model to be an oscillatory system and derive least squares algorithm to estimate the frequency parameters. We extend the solution to the general case of multiple noiseless sinusoids and express the global solution in terms of the inverse of a Toeplitz plus Hankel... View full abstract»

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  • The restoration of pitch variation defects in gramophone recordings

    Publication Year: 1993, Page(s):148 - 151
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (264 KB)

    A new algorithm is presented for the identification and restoration of time-varying pitch defects in audio signals. The problem is commonly encountered as `wow' in gramophone disc and magnetic tape recordings where motor speed variations or eccentricity in the recording process are significant. The algorithm operates in two stages, the first of which trades tonal components in musical signals to g... View full abstract»

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  • Robust adaptive processing of microphone array data for hearing aids

    Publication Year: 1993, Page(s):77 - 80
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (276 KB)

    The problem of adaptively combining the outputs of an array of microphones as a single input for a hearing aid is investigated. A robust processor based on a constrained minimum variance optimization approach is used. One fundamental criteria employed in designing this robust beamformer limits the amount of cancellation of the desired signal. The results presented include the effects of acoustic h... View full abstract»

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  • Computationally efficient compression of audio signals by means of RIQ-DPCM

    Publication Year: 1993, Page(s):35 - 38
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (256 KB)

    The need to transmit large amounts of data over limited bandwidth channels has resulted in many methods for digital data compression. The common approach is to identify and remove redundancy from the input data stream using knowledge of the source characteristics. In the case of signals intended for human observers (speech, music, pictures, etc.) it is also useful to consider the strengths and wea... View full abstract»

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  • A microphone array for multimedia applications

    Publication Year: 1993, Page(s):52 - 55
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (224 KB)

    A microphone array for speech pick-up is presented. This array, intended to be used in multimedia workstations for hands-free communication, is based on simple acoustic principles and it has been implemented with standard technology. Performance characteristics are given in terms of acoustic behaviour; moreover, results of a listening test are presented which show that speech picked up by the arra... View full abstract»

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  • Hierarchic models of hearing for sound separation and reconstruction

    Publication Year: 1993, Page(s):157 - 160
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (304 KB)

    In building a machine to detect and segregate individual components in sound mixtures, the best example to copy is the human auditory system. Several models of auditory organization implement various rules of psychoacoustic grouping. We propose in addition to model auditory inference as exhibited in the well-known `phonemic restoration illusion' of Warren (1970). A hierarchy of abstracted features... View full abstract»

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  • Hearing aids for profoundly deaf people based on a new parametric concept

    Publication Year: 1993, Page(s):89 - 92
    Cited by:  Papers (1)  |  Patents (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (252 KB)

    People with severe hearing loss only have a minor part of the frequency range available for reception of information in speech signals. These people do not benefit from normal hearing aids as the information in high frequency parts of the speech is not available. To overcome this problem the authors have developed a new method enabling to present information from the frequency range of interest in... View full abstract»

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  • Real-time generation of interactive virtual auditory environments

    Publication Year: 1993, Page(s):106 - 109
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (324 KB)

    Virtual auditory environments refer to a procedure in which auditory environments are created by means of a computer model (Lehnert & Blauert 1991). These artificial environments are perceived as being natural and they create the impression of being present in another physical space. The sense of tele-presence can greatly be improved by making these environments interactive, that is, the subje... View full abstract»

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  • Developments in transaural stereo

    Publication Year: 1993, Page(s):114 - 117
    Cited by:  Papers (27)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    Transaural stereo achieves precision 3-D imaging by compensating for spectral distortions in the loudspeaker-to-car signal paths. The heart of transaural stereo, signal processing for crosstalk cancellation, is herein generalized to accommodate any number of loudspeakers and listeners in any layout. Transaural equations are written and then solved using standard algebraic methods. Worked-out examp... View full abstract»

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  • Aspects of current standardization activities for high-quality, low-rate multi-channel audio coding

    Publication Year: 1993, Page(s):47 - 50
    Cited by:  Papers (3)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (348 KB)

    This paper analyzes directions in the current standardization activities for multi-channel audio, briefly reviews the composite coding schemes AC-3 and ISO 11172-3 compatible systems, and discusses requirements, features, and time-tables for the audio systems in the ISO/Moving pictures Expert Group (MPEG) phase 2 and the United States high definition television (HDTV) standardization processes View full abstract»

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  • A comparison of gradient-based algorithms for echo compensation with decorrelating properties

    Publication Year: 1993, Page(s):12 - 15
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    Cancelling echoes by using the normalized least mean square (NLMS) algorithm has been state of the art for many years. In acoustical echo compensation, however, it is common to estimate more than 1000 parameters resulting in a too slow convergence when driven by speech signals. In order to overcome this drawback, a lot of modifications have been published in the last years, all having one goal: to... View full abstract»

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  • Adaptive predictive coding with transform domain quantization using block size adaptation and high-resolution spectral modeling

    Publication Year: 1993, Page(s):31 - 34
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (252 KB)

    The adaptive predictive coding with transform domain quantization (APC-TQ) technique was proposed by Bhaskar (1991) for the compression of audio signals. Since then, significant developments have taken place leading to a reduction in the coding rate. While enhancing the audio quality. These developments include (i) the use of block size adaptation to exploit the variations in the stationarity of t... View full abstract»

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  • Principle and application of a new test signal to determine the transfer characteristics of telecommunication systems

    Publication Year: 1993, Page(s):152 - 155
    Cited by:  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    Measuring procedures yielding defined and reproducible results are required to determine transfer functions for tests and registrations. On the one hand, such a test signal allowing the determination of the transfer characteristics of these systems must simulate voice properties adequately. On the other hand, such a signal must be determined exactly so that not only the transfer function in differ... View full abstract»

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  • Multidimensional scaling analysis of head-related transfer functions

    Publication Year: 1993, Page(s):98 - 101
    Cited by:  Papers (1)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    Accurate rendering of auditory objects in a virtual auditory display depends on signal processing that is based on detailed measurements of the human free-field to eardrum transfer function (HRTF). The performance of an auditory display can be severely compromised if the HRTF measurements are not made individually, for each potential user. This requirement could sharply limit the practical applica... View full abstract»

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  • A new technique to measure electroacoustic transducer directivity indices in reverberant fields

    Publication Year: 1993, Page(s):64 - 67
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (268 KB)

    The paper presents a new method for measuring the directivity index of an electroacoustic transducer in a diffuse reverberant environment. The method that is proposed relies on the measurement of the spectral density variance of the transfer function between source and receiver. The method requires a measurement of the source/receiver transfer function, the distance between source and receiver, th... View full abstract»

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  • Perceptual consequences of interpolating head-related transfer functions during spatial synthesis

    Publication Year: 1993, Page(s):102 - 105
    Cited by:  Papers (7)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (324 KB)

    In implementing a spatial auditory display, many engineering compromises must be made to achieve a practical system. One such compromise involves devising methods for interpolating between the head-related transfer functions (HRTFs) used to synthesize spatial stimuli in order to achieve smooth motion trajectories and locations at finer resolutions than the empirical data. The perceptual consequenc... View full abstract»

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  • Dithered quantizers with and without feedback

    Publication Year: 1993, Page(s):140 - 143
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (236 KB)

    It is shown that quantizing systems without feedback respond to the use of particular spectrally-shaped dither signals quite differently from those with feedback paths. For each type of system, conditions are given which ensure that the quantization error will be wide-sense stationary with no input dependence and with a predictable power spectral density function View full abstract»

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  • Constraint based audio interpolators

    Publication Year: 1993, Page(s):161 - 164
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (272 KB)

    In audio digital signal processing, interpolators are used for a variety of functions, including sample rate conversion. Linear interpolation is commonly used, but has serious signal quality problems for signals with significant high frequency content. Higher order interpolators based on sine functions or other conventional lowpass filter design techniques offer somewhat better performance, but ar... View full abstract»

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  • Current and future standardization of high-quality digital audio coding in MPEG

    Publication Year: 1993, Page(s):43 - 46
    Cited by:  Papers (3)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (340 KB)

    Since 1988 ISO/IEC JTCI/SC29 WG11 (MPEG) is working on the standardization of video and audio signals. The Audio subgroup of MPEG is working on bit rate reduction systems for high quality digital audio. Since the first phase of this standardization effort has been finished, MPEG/Audio is extending its work to multichannel audio coding systems as well as to medium quality coding at lower sampling f... View full abstract»

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