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Experience with speech communication in packet networks

December 1, 1983
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Journal Article
Author:
Clifford J. Weinstein
…
Published in:
IEEE J. Sel. Areas Commun., Vol. SAC-1, No. 6, December 1983, pp. 963-980.
Topic:
signal processing
R&D area:
R&D group:

Summary

The integration of digital voice with data in a common packet-switched network system offers a number of potential benefits, including reduced systems cost through sharing of switching and transmission resources, flexible internetworking among systems utilizing different transmission media, and enhanced services for users requiring access to both voice and data communications. Issues which it has been necessary to address in order to realize these benefits include reconstitution of speech from packets arriving at nonuniform intervals, maximization of packet speech multiplexing efficiency, and determination of the implementation requirements for terminals and switching in a large-scale packet voice/data system. A series of packet speech systems experiments to address these issues has been conducted under the sponsorship of the Defense Advanced Research Projects Agency (DARPA). In the initial experiments on the ARPANET, the basic feasibility of speech communication on a store-and-forward packet network was demonstrated. Techniques were developed for reconstitution of speech from packets, and protocols were developed for call setup and for speech transport. Later speech experiments utilizing the Atlantic packet satellite network (SATNET) led to the development of techniques for efficient voice conferencing in a broadcast environment, and for internetting speech between a store-and-forward net (ARPANET) and a broadcast net (SATNET). Large-scale packet speech multiplexing experiments could not be carried out on ARPANET or SATNET where the network link capacities severely restrict the number of speech users that can be accommodated. However, experiments are currently being carried out using a wide-band satellite-based packet system designed to accommodate a sufficient number of simultaneous users to support realistic experiments in efficient statistical multiplexing. Key developments to date associated with the wide-band experiments have been 1) techniques for internetting via voice/data gateways from a variety of local access networks (packet cable, packet radio, and circuit-switched) to a long-haul broadcast satellite network and 2) compact implementations of packet voice terminals with full protocol and voice capabilities. Basic concepts and issues associated with packet speech systems are described. Requirements and techniques for speech processing, voice protocols, packetization and reconstitution, conferencing, and multiplexing are discussed in the context of a generic packet speech system configuration. Specific experimental configurations and key packet speech results on the ARPANET, SATNET, and wide-band system are reviewed.
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Summary

The integration of digital voice with data in a common packet-switched network system offers a number of potential benefits, including reduced systems cost through sharing of switching and transmission resources, flexible internetworking among systems utilizing different transmission media, and enhanced services for users requiring access to both voice and data...

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Experience with speech communication in packet networks

Frequency sampling of the short-time Fourier-transform magnitude for signal reconstruction

November 1, 1983
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Journal Article
Author:
Jae S. Lim
…
Published in:
J. Opt. Soc. Amer., Vol. 73, November 1983, pp. 1523- 1526.
Topic:
signal processing
R&D area:
R&D group:

Summary

Unique recovery of a signal from the magnitude (modulus) of the Fourier transform has been of long-standing interest in image and optical processing in which Fourier-transform phase is lost or difficult to measure. We investigate an alternative problem of recovering a signal from the Fourier-transform magnitude of overlapping regions of the signal, i.e., from the short-time (or -space) Fourier-transform magnitude. Recently it was established that a discrete-time signal x (n) can be uniquely obtained under mild restrictions from its short-time Fourier-transform magnitude. In this paper we extend this result to the case when the short-time Fourier-transform magnitude is known at only one or two frequencies for each n. We also present a recursive algorithm for recovering a sequence from such samples and demonstrate the algorithm with an example.
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Summary

Unique recovery of a signal from the magnitude (modulus) of the Fourier transform has been of long-standing interest in image and optical processing in which Fourier-transform phase is lost or difficult to measure. We investigate an alternative problem of recovering a signal from the Fourier-transform magnitude of overlapping regions of...

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Frequency sampling of the short-time Fourier-transform magnitude for signal reconstruction

The Experimental Integrated Switched Network - a system-level network test facility

October 31, 1983
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Conference Paper
Author:
Clifford J. Weinstein
Published in:
Proc. 1983 IEEE Military Communications Conf., MILCOM, 31 October-2 November 1983.
Topic:
signal processing
R&D area:
R&D group:

Summary

An Experimental Integrated Switched Network (EISN) has been developed to provide a system-level testbed for the evaluation of advanced communications networking techniques, including survivable network routing algorithms using a mix of transmission media, for application in the Defense Switched Network (DSN). EISN includes five CONUS sites linked by a wideband demand-assigned satellite channel and by dialed-up terrestrial trunks for alternate satellite/terrestrial routing experiments. Experiments to date have validated techniques for integration of circuit-switched terrestrial systems with the demand-assigned satellite system, and for the establishment of alternate routes over satellite and terrestrial paths. Currently, candidate routing algorithms for application in the DSN are being implemented and tested using external routing/controller processors attached to digital circuit switches at EISN sites. In addition, EISN is also being used to support data communication experiments using DoD standard data protocols in a combined satellite/terrestrial network environment. Work is ongoing both in system experiments and in testbed developments to include additional capabilities. This paper represents a description and status report on both the testbed and the experimental efforts.
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Summary

An Experimental Integrated Switched Network (EISN) has been developed to provide a system-level testbed for the evaluation of advanced communications networking techniques, including survivable network routing algorithms using a mix of transmission media, for application in the Defense Switched Network (DSN). EISN includes five CONUS sites linked by a wideband...

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The Experimental Integrated Switched Network - a system-level network test facility

Object detection by two-dimensional linear prediction

January 28, 1983
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Technical Report
Author:
Thomas F. Quatieri
Published in:
MIT Lincoln Laboratory Report TR-632
Topic:
image processing
R&D area:
R&D group:

Summary

An important component of any automated image analysis system is the detection and classification of objects. In this report, we consider the first of these problems where the specific goal is to detect anomalous areas (e.g., man-made objects) in textured backgrounds such as trees, grass, and fields of aerial photographs. Our detection algorithm relies on a significance test which adapts itself to the changing background in such a way that a constant false alarm rate is maintained. Furthermore, this test has a potentially practical implementation since it can be expressed in terms of the residuals of an adaptive two-dimensional linear predictor. The algorithm is demonstrated with both synthetic and realworld images.
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Summary

An important component of any automated image analysis system is the detection and classification of objects. In this report, we consider the first of these problems where the specific goal is to detect anomalous areas (e.g., man-made objects) in textured backgrounds such as trees, grass, and fields of aerial photographs...

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Object detection by two-dimensional linear prediction

Implementation of 2-D digital filters by iterative methods

June 1, 1982
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Journal Article
Author:
Thomas F. Quatieri
…
Published in:
IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-30, No. 3, June 1982, pp. 473-87.
Topic:
signal processing
R&D area:
R&D group:

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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Implementation of 2-D digital filters by iterative methods

Signal reconstruction from the short-time Fourier transform magnitude

May 2, 1982
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Conference Paper
Author:
Jae S. Lim
…
Published in:
IEEE-ASSP Int. Conf., 2 May 1982.
Topic:
signal processing
R&D area:
R&D group:

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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Signal reconstruction from the short-time Fourier transform magnitude

Iterative techniques for minimum phase signal reconstruction from phase or magnitude

December 1, 1981
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Journal Article
Author:
Thomas F. Quatieri
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Published in:
IEEE Trans. on Acoustics, Speech & Signal Processing, Vol. ASSP-29, No.6, Dec. 1981, pp.1187-1193.
Topic:
speech enhancement
R&D area:
R&D group:

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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Iterative techniques for minimum phase signal reconstruction from phase or magnitude

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

May 1, 1981
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Journal Article
Author:
Thomas F. Quatieri
Published in:
Proc. IEEE, Vol. 69, No. 5, May 1981, pp. 667-668.
Topic:
signal processing
R&D area:
R&D group:

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

The effects of microphones and facemasks on LPC vocoder performance

March 1, 1981
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Conference Paper
Author:
Elliot Singer
Published in:
Proc. of IEEE Int. Conf. on Acoustics, Speech & Signal Processing, 30 March - 1 April 1981.
Topic:
speech
R&D area:

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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The effects of microphones and facemasks on LPC vocoder performance

Voice communication in integrated digital voice and data networks

September 1, 1980
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Journal Article
Author:
Theodore Bially
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Published in:
IEEE Trans. Commun., Vol. COM-28, No. 9, September 1980, pp. 1478-90.
Topic:
signal processing
R&D area:
R&D group:

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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Voice communication in integrated digital voice and data networks
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