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Automatic talker activity labeling for co-channel talker interference suppression
March 21, 1990
Conference Paper
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, Vol. 2, Speech Processing 2; VLSI; Audio and Electroacoustics, ICASSP, 3-6 April 1990, pp. 813-816.
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Summary
This paper describes a speaker activity detector taking co-channel speech as input and labeling intervals of the input as target-only, jammer-only, or two-speaker (target+jammer). The algorithms applied were borrowed primarily from speaker recognition, thereby allowing us to use speaker-dependent test-utterance-independent information in a front-end for co-channel talker interference suppression. Parameters studied included classifier choice (vector quantization vs. Gaussian), training method (unsupervised vs. supervised), test utterance segmentation (uniform vs. adaptive), and training and testing target-to-jammer ratios. Using analysis interval lengths of 100 ms, performance reached 80% correct detection.
Summary
This paper describes a speaker activity detector taking co-channel speech as input and labeling intervals of the input as target-only, jammer-only, or two-speaker (target+jammer). The algorithms applied were borrowed primarily from speaker recognition, thereby allowing us to use speaker-dependent test-utterance-independent information in a front-end for co-channel talker interference suppression. Parameters...
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Microburst detection with airport surveillance radars
August 1, 1989
Conference Paper
Published in:
34th Ann. Air Traffic Control Associsation., 30 October 1989 - 2 November 1989, pp. 514-522.
Summary
With the advent of fully digital signal processing for new airport surveillance radars (ASR-9), terminal air traffic control displays will be largely free of clutter from precipitation and ground scatterers [1,2]. Early acceptance testing of the ASR-9, however, indicated that working air traffic controllers actually made considerable use of the weather echo information on their displays. To reinsert weather data in a non-interfering manner, the ASR-9's signal processor was augmented with a dedicated channel for processing and displaying six quantitative levels of precipitation reflectivity (i.e. rain rate) [2,3]. This processor does not utilize tile radar's coherency, other than for Doppler filtering of ground clutter echoes. In this paper, we describe processing techniques that would allow airport surveillance radars to extend their weather measurement capability to the detection of microburst-generated low altitude wind shear. The two principal technical challenges are the development of (i) signal processing to suppress ground clutter and estimate the near surface radial wind component in each radar resolution cell; (ii) image processing to automatically detect hazardous shear in the resulting velocity field. The techniques have been evaluated extensively using simulated weather signals and measurements from an experimental airport surveillance radar in the southeastern United States. Overall our analysis indicates that microbursts accompanied by rain at the surface -- the predominant safety hazard in many parts of the U.S. --could be detected with high confidence using a suitably modified ASR. In the following section we describe briefly the background and potential operational role of an ASR-based wind shear detection system. We then discuss the primary technical issues for achieving this capability and our evaluations of processing methods that address these issues.
Summary
With the advent of fully digital signal processing for new airport surveillance radars (ASR-9), terminal air traffic control displays will be largely free of clutter from precipitation and ground scatterers [1,2]. Early acceptance testing of the ASR-9, however, indicated that working air traffic controllers actually made considerable use of the...
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Speech-state-adaptive simulation of co-channel talker interference suppression
May 23, 1989
Conference Paper
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, 23-26 May 1989, pp. 361-364.
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Summary
A co-channel talker interference suppression system processes an input waveform containing the sum of two simultaneous speech signals, referred to as the target and the jammer, to produce a waveform estimate of the target speech signal alone. This paper describes the evaluation of a simulated suppression system performing ideal suppression of a jammer signal given the voicing states (voiced, unvoiced, silent) of the target and jammer speech as a function of time and given the isolated target and jammer speech waveforms. By applying suppression to select regions of jammer speech as a function of the voicing states of the target and jammer, and by measuring the intelligibility of the resulting jammer suppressed co-channel speech, it is possible to identify those regions of co-channel speech on which interference suppression most improves intelligibility. Such results can help focus algorithm development efforts.
Summary
A co-channel talker interference suppression system processes an input waveform containing the sum of two simultaneous speech signals, referred to as the target and the jammer, to produce a waveform estimate of the target speech signal alone. This paper describes the evaluation of a simulated suppression system performing ideal suppression...
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A block diagram compiler for a digital signal processing MIMD computer
April 9, 1987
Conference Paper
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 4, 6-9 April 1987, pp. 1867-1870.
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Summary
A Block Diagram Compiler (BOC) has been designed and implemented for converting graphic block diagram descriptions of signal processing tasks into source code to be executed on a Multiple Instruction Stream - Multiple Data Stream (MIMD) array computer. The compiler takes as input a block diagram of a real-time DSP application, entered on a graphics CAE workstation, and translates it into efficient real-time assembly language code for the target multiprocessor array. The current implementation produces code for a rectangular grid of Texas Instruments TMS32010 signal processors built at Lincoln Laboratory, but the concept could be extended to other processors or other geometries in the same way that a good assembly language programmer would write it. This report begins by examining the current implementation of the BOC including relevant aspects of the target hardware. Next, we describe the task-assignment module, which uses a simulated annealing algorithm to assign the processing tasks of the DSP application to individual processors in the array. Finally, our experiences with the current version of the BOC software and hardware are reported.
Summary
A Block Diagram Compiler (BOC) has been designed and implemented for converting graphic block diagram descriptions of signal processing tasks into source code to be executed on a Multiple Instruction Stream - Multiple Data Stream (MIMD) array computer. The compiler takes as input a block diagram of a real-time DSP...
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Mixed-phase deconvolution of speech based on a sine-wave model
April 9, 1987
Conference Paper
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 2, 6-9 April 1987, pp. 649-652.
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Summary
This paper describes a new method of deconvolving the vocal cord excitation and vocal tract system response. The technique relies on a sine-wave representation of the speech waveform and forms the basis of an analysis-synthesis method which yields synthetic speech essentially indistinguishable from the original. Unlike an earlier sinusoidal analysis-synthesis technique that used a minimum-phase system estimate, the approach in this paper generates a "mixed-phase" system estimate and thus an improved decomposition of excitation and system components. Since a mixed-phase system estimate is removed from the speech waveform, the resulting excitation residual is less dispersed than the previous sinusoidal-based excitation estimate of the more commonly used linear prediction residual. A method of time-varying linear filtering is given as an alternative to sinusoidal reconstruction, similar to conventional time-domain synthesis used in certain vocoders, but without the requirement of pitch and voicing decisions. Finally, speech modification with a mixed-phase system estimate is shown to be capable of more closely preserving waveform shape in time-scale and pitch transformations than the earlier approach.
Summary
This paper describes a new method of deconvolving the vocal cord excitation and vocal tract system response. The technique relies on a sine-wave representation of the speech waveform and forms the basis of an analysis-synthesis method which yields synthetic speech essentially indistinguishable from the original. Unlike an earlier sinusoidal analysis-synthesis...
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Experience with speech communication in packet networks
December 1, 1983
Journal Article
Published in:
IEEE J. Sel. Areas Commun., Vol. SAC-1, No. 6, December 1983, pp. 963-980.
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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 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.
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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Frequency sampling of the short-time Fourier-transform magnitude for signal reconstruction
November 1, 1983
Journal Article
Published in:
J. Opt. Soc. Amer., Vol. 73, November 1983, pp. 1523- 1526.
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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 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.
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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The Experimental Integrated Switched Network - a system-level network test facility
October 31, 1983
Conference Paper
Published in:
Proc. 1983 IEEE Military Communications Conf., MILCOM, 31 October-2 November 1983.
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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 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.
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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Implementation of 2-D digital filters by iterative methods
June 1, 1982
Journal Article
Published in:
IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-30, No. 3, June 1982, pp. 473-87.
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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 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.
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
May 2, 1982
Conference Paper
Published in:
IEEE-ASSP Int. Conf., 2 May 1982.
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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 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.
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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