Scheduled System Maintenance
On Friday, October 20, IEEE Xplore will be unavailable from 9:00 PM-midnight ET. We apologize for the inconvenience.

# Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

## Filter Results

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

Publication Year: 1993
| PDF (43 KB)
• ### Dithered quantizers with and without feedback

Publication Year: 1993, Page(s):140 - 143
Cited by:  Papers (5)
| | PDF (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»

• ### Robust adaptive processing of microphone array data for hearing aids

Publication Year: 1993, Page(s):77 - 80
Cited by:  Papers (2)
| | PDF (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»

• ### Detection and restoration of sound of flute embedded in noise using real-time Kalman filter

Publication Year: 1993, Page(s):144 - 147
| | PDF (236 KB)

The restoration of flute notes embedded in noise is formulated as a state-estimation problem of a dynamic system. A single Kalman filter, with a given state-transition matrix, is implemented in real-time to recover the corresponding note as well as some of the neighbouring notes. In order to restore a continuous piece of music played by flute, a bank of Kalman filters (with different state-transit... View full abstract»

• ### A simplified source/filter model for percussive sounds

Publication Year: 1993, Page(s):173 - 176
Cited by:  Papers (1)
| | PDF (352 KB)

This paper deals with source-filter models of percussive instruments. A multi-channel excitation/filter model' is presented in which a single excitation is used to generate several sounds, for example six piano tones belonging to the same octave. Techniques for estimating the model parameters are presented and applied to the sound of a real piano. Our experiments demonstrate that it is possible t... View full abstract»

• ### Robust real-time constrained hearing aid arrays

Publication Year: 1993, Page(s):81 - 84
Cited by:  Papers (1)  |  Patents (4)
| | PDF (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»

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

Publication Year: 1993, Page(s):64 - 67
| | PDF (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»

• ### 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)
| | PDF (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»

• ### The restoration of pitch variation defects in gramophone recordings

Publication Year: 1993, Page(s):148 - 151
Cited by:  Papers (2)
| | PDF (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»

• ### The 2-D digital waveguide mesh

Publication Year: 1993, Page(s):177 - 180
Cited by:  Papers (47)
| | PDF (256 KB)

An extremely efficient method for modeling wave propagation in a membrane is provided by the multidimensional extension of the digital waveguide. The 2-D digital waveguide mesh is constructed out of bi-directional delay units and scattering junctions. We show that it coincides with the standard finite difference scheme in the lossless case. Wave propagation in the mesh is compared with wave propag... View full abstract»

• ### A time-frequency neutral network layered model for hearing perception

Publication Year: 1993, Page(s):123 - 126
Cited by:  Papers (1)
| | PDF (260 KB)

This paper introduces a layered neural network model for hearing perception. It is based on five important perceptual properties of hearing. The neural network model processes a joint-domain representation of the input signal to yield the desired perceptual properties. The focus is on the first two layers of the model, the transformation layer and two feature extraction layers View full abstract»

• ### An all digital concha hearing aid

Publication Year: 1993, Page(s):85 - 88
Cited by:  Papers (1)
| | PDF (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»

• ### Local silencing of room acoustic noise using broadband active noise control

Publication Year: 1993, Page(s):23 - 25
Cited by:  Papers (4)
| | PDF (188 KB)

Adaptive filtering techniques are now in widespread use for a number of applications such as adaptive arrays, adaptive noise cancellation, adaptive line enhancement, adaptive modeling and system identification, adaptive equation, and adaptive echo cancellation. These techniques have also been applied to the expanding field of active noise control. In this paper, an application of active noise cont... View full abstract»

• ### Parametric approximation of room impulse responses based on wavelet decomposition

Publication Year: 1993, Page(s):68 - 71
Cited by:  Papers (1)  |  Patents (2)
| | PDF (240 KB)

A new approach to the approximation and real-time simulation of room impulse responses is presented. Based on wavelet decomposition of measured impulse response data an energy-time-frequency representation of the system room is obtained. The wavelet coefficients in the frequency subbands are calculated by a multirate analysis filter bank providing aliasing-free subband processing and linear-phase ... View full abstract»

• ### Directional microphones in computer simulated and real rooms

Publication Year: 1993, Page(s):56 - 59
| | PDF (216 KB)

The subjective effects of utilizing highly directional microphones in a teleconferencing setting are not well understood. Computer simulation of both complex microphone systems and room environments offer one opportunity to study the combined effects. A complex microphone system can be modeled as a collection of point microphones distributed in space and summed with appropriate time delays. Establ... View full abstract»

• ### 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)
| | PDF (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»

• ### 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)
| | PDF (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»

• ### Time-scale modification with temporal envelope invariance

Publication Year: 1993, Page(s):127 - 130
Cited by:  Papers (1)
| | PDF (316 KB)

A new approach is introduced for time-scale modification of short-duration complex acoustic signals to improve their audibility. The method preserves the time-scaled temporal envelope of a signal and for enhancement capitalizes on the perceptual importance of a signal's temporal structure. The basis for the approach is a sub-band representation whose channel phases are controlled to shape the the ... View full abstract»

• ### Constraint based audio interpolators

Publication Year: 1993, Page(s):161 - 164
Cited by:  Papers (1)  |  Patents (1)
| | PDF (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»

• ### 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)
| | PDF (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»

• ### Interpolation of forced structural responses from non-uniform sparse measurements

Publication Year: 1993, Page(s):26 - 29
| | PDF (264 KB)

This paper presents a method for interpolating a sparse set of nonuniformly spaced velocity measurements on the surface of a vibrating structure. The method utilizes knowledge of the physical nature of the vibrating structure specified in terms of a given bound on the energy of the excitation forces, estimated mobilities of the structure and a known set of sparse velocity measurements. To minimize... View full abstract»

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

Publication Year: 1993, Page(s):102 - 105
Cited by:  Papers (13)  |  Patents (1)
| | PDF (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»

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

Publication Year: 1993, Page(s):136 - 139
Cited by:  Papers (1)
| | PDF (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»

• ### Superdirective arrays for hearing aids

Publication Year: 1993, Page(s):73 - 76
| | PDF (220 KB)

Microphone arrays are the most effective of the techniques that have been proposed for improving speech intelligibility in noise for the hearing impaired. Superdirective arrays are attractive since optimal performance can be obtained for a stationary random noise field. A constrained superdirective array suitable for hearing-aid applications is discussed in the paper View full abstract»

• ### HNM: a simple, efficient harmonic+noise model for speech

Publication Year: 1993, Page(s):169 - 172
Cited by:  Papers (11)  |  Patents (6)
| | PDF (336 KB)

HNM, a new analysis/modification/synthesis model based on a harmonic+noise representation of the speech signal is presented. The HNM model has several specificities: (1) HNM assumes the speech signal to be composed of a deterministic part and of a stochastic part, (2) the deterministic part is assumed to contain only harmonically related sinusoids with linearly varying complex amplitudes, and (3) ... View full abstract»