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Implementation of 2-D digital filters by iterative methods

Published in:
IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-30, No. 3, June 1982, pp. 473-87.

Summary

A two-dimensional (2-D) rational filter can be implemented by an iterative computation involving only finite-extent impulse response (FIR) filtering operations, provided a certain convergence criterion is met. In this paper, we generalize this procedure so that the convergence criterion is satisfied for any stable 2-D rational transfer function. One formulation which guarantees convergence invokes a relaxed form of the iterative computation along with prefiltering the numerator and denominator polynomials of the rational transfer function. This implementation may be applied with a frequency-varying relaxation parameter for increasing the rate of convergence. An alternative generalization uses several previously computed iterates, unlike our first modification which utilizes only the most recently computed iterate. This formulation can potentially guarantee convergence and also increase the convergence rate without the requirement of prefiltering. Another extension of the iterative computation incorporates constraints (e.g., positivity or finite extent) on the output of each iteration. Proof of convergence of such constrained iterations relies on the concept of a nonexpansive operator. In particular, the error introduced within the converging solution resulting from a finite-extent constraint is shown to satisfy a homogeneous partial difference equation. Finally, this error computation leads to an important link between our iterative implementation with constraints and an iterative solution to partial difference equations (e.g., Laplace's equation) with known boundary conditions.
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Summary

A two-dimensional (2-D) rational filter can be implemented by an iterative computation involving only finite-extent impulse response (FIR) filtering operations, provided a certain convergence criterion is met. In this paper, we generalize this procedure so that the convergence criterion is satisfied for any stable 2-D rational transfer function. One formulation...

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Signal reconstruction from the short-time Fourier transform magnitude

Published in:
IEEE-ASSP Int. Conf., 2 May 1982.

Summary

In this paper, a signal is shown to be uniquely represented by the magnitude of its short-time Fourier transform (STFT) under mild restrictions on the signal and the analysis window of the STFT. Furthermore, various algorithms are developed which reconstruct signal from appropriate samples of the STFT magnitude. Several of the algorithms can also be used to obtain signal estimates from the processed STFT magnitude, which generally does not have a valid short-time structure. These algorithms are successfully applied to the time-scale modification and noise reduction problems in speech processing. Finally, the results presented here have similar potential for other applications areas, including those with multidimensional signals.
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Summary

In this paper, a signal is shown to be uniquely represented by the magnitude of its short-time Fourier transform (STFT) under mild restrictions on the signal and the analysis window of the STFT. Furthermore, various algorithms are developed which reconstruct signal from appropriate samples of the STFT magnitude. Several of...

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Iterative techniques for minimum phase signal reconstruction from phase or magnitude

Published in:
IEEE Trans. on Acoustics, Speech & Signal Processing, Vol. ASSP-29, No.6, Dec. 1981, pp.1187-1193.

Summary

In this paper, we develop iterative algorithms for reconstructing a minimum phase sequence from pthhea se or magnitude of its Fourier transform. These iterative solutions involve repeatedly imposing a causality constraint in the time domain and incorporating the known phase or magnitude function in the frequency domain. This approach is the basis of a new means of computing the Hilbert transform of the log-magnitude or phase of the Fourier transform of a minimum phase sequence which does not require phase unwrapping. Finally, we discuss the potential use of this iterative computation in determining samples of the unwrapped phase of a mixed phase sequence.
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Summary

In this paper, we develop iterative algorithms for reconstructing a minimum phase sequence from pthhea se or magnitude of its Fourier transform. These iterative solutions involve repeatedly imposing a causality constraint in the time domain and incorporating the known phase or magnitude function in the frequency domain. This approach is...

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Recursive two-dimensional signal reconstruction from linear system input and output magnitudes

Published in:
Proc. IEEE, Vol. 69, No. 5, May 1981, pp. 667-668.

Summary

A recursive algorithm is presented for reconstructing a two-dimensional complex signal from its magnitude and the magnitude of the output of a known linear shift-invariant system whose input is the desired signal. The recursion has a simple geometric interpretation, and is easily extended to causal, shift-varying systems.
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Summary

A recursive algorithm is presented for reconstructing a two-dimensional complex signal from its magnitude and the magnitude of the output of a known linear shift-invariant system whose input is the desired signal. The recursion has a simple geometric interpretation, and is easily extended to causal, shift-varying systems.

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The effects of microphones and facemasks on LPC vocoder performance

Author:
Published in:
Proc. of IEEE Int. Conf. on Acoustics, Speech & Signal Processing, 30 March - 1 April 1981.

Summary

The effects of oxygen facemasks and noise cancelling microphones on LPC vocoder performance were analyzed and evaluated. Likely sources of potential vocoder performance degradation included the non-ideal frequency response characteristics of the microphone and the possible presence of additional resonances in the speech waveform due to the addition of the facemask cavity. Examination of vowel spectra revealed that spurious resonances do not occur in the vocoder frequency band for speech generated using the facemask and microphone. Also observed was a vowel-dependent reduction in the bandwidths of the upper formants, a result which can be predicted from acoustic theory. Finally, it is shown that the low frequency emphasis associated with small enclosures is not relevant when using a pressure gradient (noise cancelling) microphone. Diagnostic Rhyme Tests involving three subjects indicated that the presence of the oxygen facemask and noise cancelling microphone did not result in a significant increase in the LPC vocoder processing loss.
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Summary

The effects of oxygen facemasks and noise cancelling microphones on LPC vocoder performance were analyzed and evaluated. Likely sources of potential vocoder performance degradation included the non-ideal frequency response characteristics of the microphone and the possible presence of additional resonances in the speech waveform due to the addition of the...

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Voice communication in integrated digital voice and data networks

Published in:
IEEE Trans. Commun., Vol. COM-28, No. 9, September 1980, pp. 1478-90.

Summary

Voice communication networks have traditionally been designed to provide either analog signal paths or fixed-rate synchronous digital connections between individual subscribers. These designs were aimed at accommodating the "streamlike" character of speech, which has traditionally been considered to flow from source to destination at a more or less constant rate. By way of contrast, interactive and computer-to-computer data transactions tend to be "bursty" in nature, and this has given rise to the development of packet-switching methods for data communications. The dichotomous nature of these two major traffic classes and the apparent conflict between the types of network services they require has resulted in the deployment of separate military communications facilities for voice and data. A challenge in the design of future systems is to achieve overall economy and flexibility in the allocation of resources via the efficient integration of both traffic types in common network facilities. This paper summarizes a number of advanced concepts for switching and flow control of combined voice and data traffic in integrated environments. Performance characteristics are described based on analysis results and computer simulation studies for both multilink terrestrial and broadcast satellite network topologies.
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Summary

Voice communication networks have traditionally been designed to provide either analog signal paths or fixed-rate synchronous digital connections between individual subscribers. These designs were aimed at accommodating the "streamlike" character of speech, which has traditionally been considered to flow from source to destination at a more or less constant rate...

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Convergence of iterative nonexpansive signal reconstruction algorithms

Published in:
IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-29, No. 5, October 1981, pp. 1052-1059.

Summary

Iterative algorithms for signal reconstruction from partial time- and frequency-domain knowledge have proven useful in a number of application areas. In this paper, a general convergence proof, applicable to a general class of such iterative reconstruction algorithms, is presented. The proof relies on the concept of a nonexpansive mapping in both the time and frequency domains. Two examples studied in detail are time-limited extrapolation (equivalently, band-limited extrapolation) and phase-only signal reconstruction. The proof of convergence for the phase-only iteration is a new result obtained by this method of proof. The generality of the approach allows the incorporation of nonlinear constraints such as time- (or space-) domain positivity or minimum and maximum value constraints. Finally, the underrelaxed form of these iterations is also shown to converge even when the solution is not guaranteed to be unique.
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Summary

Iterative algorithms for signal reconstruction from partial time- and frequency-domain knowledge have proven useful in a number of application areas. In this paper, a general convergence proof, applicable to a general class of such iterative reconstruction algorithms, is presented. The proof relies on the concept of a nonexpansive mapping in...

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Data traffic performance of an integrated circuit and packet-switched multiplex structure

Published in:
IEEE Trans. on Commun., Vol. COM-28, No. 6, June 1980, pp. 873-878.

Summary

Results are developed for data traffic performance in an integrated multiplex structure which includes circuit-switching for voice and packet-switching for data. The results are obtained both through simulation and analysis, and show that excessive data queues and delays will build up under heavy loading conditions. These large data delays occur during periods of time when the voice traffic load through the multiplexer exceeds its statistical average. A variety of flow control mechanisms to reduce data packet delays are investigated. These mechanisms include control of voice bit rate, limitation of the data buffer, and combinations of voice rate and data buffer control. Simulations indicate that these flow control mechanisms provide substantial improvements in system performance.
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Summary

Results are developed for data traffic performance in an integrated multiplex structure which includes circuit-switching for voice and packet-switching for data. The results are obtained both through simulation and analysis, and show that excessive data queues and delays will build up under heavy loading conditions. These large data delays occur...

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A split band adaptive predictive coding (SBAPC) speech system

Published in:
IEEE Int. Conf. on Acoustics, Speech, & Signal Processing, 9-11 April 1980.

Summary

As developed by Atal and Schroeder [1], conventional Adaptive Predictive Coding (APC) of speech employs both vocal tract and pitch prediction to achieve a low energy, spectrally flattened residual. Errors in the pitch predictor can result in clipping errors which can propagate in the system for relatively long periods of time and degrade the quality of the synthesized speech. Makhoul and Berouti [2] have developed a high quality 16 kbps APC system which eliminates the pitch predictor by using a multi-level variable rate quantizer. In order to achieve comparable quality at even lower data rates, a split band APC (SBAPC) structure is proposed which employs the multi-level quantizer on the low frequency portion of the residual and a 1-bit quantizer on the high frequency portion of the residual.
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Summary

As developed by Atal and Schroeder [1], conventional Adaptive Predictive Coding (APC) of speech employs both vocal tract and pitch prediction to achieve a low energy, spectrally flattened residual. Errors in the pitch predictor can result in clipping errors which can propagate in the system for relatively long periods of...

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The tradeoff between delay and TASI advantage in a packetized speech multiplexer

Published in:
IEEE Trans. on Commun., Vol. COM-27, No. 11, November 1979, pp. 1716-20.

Summary

A packetized speech multiplexer differs from a circuit-switched TASI system in that the presence of a packet buffer allows a tradeoff where the TASI advantage can be increased at a cost in packet delay. This tradeoff is investigated via a simulation. Results are presented to show the relations between TASI advantage and delay, for both an average delay criterion and a maximum delay criterion. It is shown that, particularly for the case where small numbers of talkers are multiplexed, the packetized system offers significant improvements in TASI advantage over the conventional circuit-switched multiplexer, at modest costs in packet delay.
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Summary

A packetized speech multiplexer differs from a circuit-switched TASI system in that the presence of a packet buffer allows a tradeoff where the TASI advantage can be increased at a cost in packet delay. This tradeoff is investigated via a simulation. Results are presented to show the relations between TASI...

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