Contains research objectives and summary of research on seven research projects.Joint Services Electronics Program (Contract DAAB07-76-C-1400)U. S. Navy - Office of Naval Research (Contract N00014-75-C-0951-NR 049-308)National Science Foundation (Grant ENG71-02319-AO2

Chan, David S. K.

Harris, David B.

Lim, Jae S.

McClellan, James H.

Oppenheim, Alan V.

Portnoff, Michael R.

Quatieri, Thomas F., Jr.

Singer, Elliot

Tribolet, José M.

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XVIII. DIGITAL SIGNAL PROCESSINGAcademic Research StaffProf. Alan V. OppenheimProf. James H. McClellanGraduate StudentsBir Bhanu Gary E. Kopec Thomas F. Quatieri, Jr.Patrick W. Bosshart Jae S. Lim Antonio RuizDavid S. K. Chan Michael R. Portnoff Elliot SingerDavid B. Harris Jos6 M. Tribolet1. TWO-DIMENSIONAL DIGITAL FILTER STRUCTURESJoint Services Electronics Program (Contract DAAB07-76-C-1400)National Science Foundation (Grant ENG71-02319-AO2)David S. K. Chan, James H. McClellanThe objective of this research is to develop insight into two-dimensional digital filterstructures by synthesizing new structures and comparing them with existing structures.Such an undertaking is important for efficient implementation of 2-D filters and in thedesign of special structures that might permit real-time processing or distributed pro-cessing.Contrary to the effect in the one-dimensional case, the signal flow graph has beenfound inadequate for representing 2-D filter structures. It fails to characterize howdata are sequenced through a structure. Information of this kind is important becausethe order in which computations are performed can seriouly affect factors in a structuresuch as storage requirements and precedence relations.We have been developing a more general framework for characterizing filter struc-tures, and our work has resulted in a new representation based on a state-spaceapproach. 1 This representation incorporates in the description of a structure the orderin which data are computed, and, in a natural manner, also describes precedencerelations between operations, an intrinsic part of any filter implementation. Such pre-cedence relations characterize the inherent limitations of a structure with regard toparallel processing, and hence are important in considering such issues as pipeliningand space/time tradeoffs.This new representation also casts light on the I-D filter implementation problem,and we are now investigating ways in which it may be applied to analysis and synthesisof I-D and 2-D digital filter structures.PR No. 119(XVIII. DIGITAL SIGNAL PROCESSING)References1. D. S. K. Chan, "A Novel Framework for the Description of Realization Structuresfor I-D and 2-D Digital Filters," EASCON '76 Record, IEEE Electronics and Aero-space Systems Convention, Washington, D. C., September 26-29, 1976, pp. 157A-157H.2. J. H. McClellan and D. S. K. Chan, "A New Structure for 2-D FIR Filters Designedby Transformations," Tenth Annual Asilomar Conference on Circuits, Systems, andComputers, Pacific Grove, California, November 22-24, 1976.2. RECONSTRUCTION OF VELOCITY STRUCTURES FROMTELESEISMIC FIRST ARRIVAL TIMESU. S. Navy- Office of Naval Research (Contract N00014-75-C-0951-NR 049-308)National Science Foundation (Grant ENG71-02319-A02)David B. Harris, James H. McClellanMany geophysical investigations require a knowledge of the spatial variation of thevelocity of wave propagation in the earth's crust and upper mantle. Some aspects ofearthquake prediction and determination of tectonic features fall into this category.Information about velocity structure on this scale is only available from measurementsmade at the earth's surface, for example, from measurements of the first arrivaltimes of teleseismic P-waves recorded at an array of seismometers.The purpose of this project is to determine the feasibility of applying the theory ofreconstruction of functions from their projections to the problem of reconstruction ofvelocity structures from recorded first arrival times. The basis of this idea is thatthe arrival time of a P-wave at a seismometer is given by a line integral of the recip-rocal velocity function along a path terminating at the seismometer. Thus, the firstarrival times recorded at an array of seismometers constitute approximately a pro-jection of the inverse velocity function onto the surface. Multiple projections from P-waves at different angles of incidence can be used to reconstruct the Fourier transformof the reciprocal velocity function. This may be inverse transformed to compute thevelocity structure.An implicit assumption of this technique is that the incoming waves have plane-wavestructure. The effects of deviations from plane-wave structure on the quality of recon-structions are being investigated by using theoretically computed first arrival times.PR No. 119(XVIII. DIGITAL SIGNAL PROCESSING)3. APPLICATION OF HOMOMORPHIC FILTERING TO SEISMICDATA PROCESSINGU.S. Navy- Office of Naval Research (Contract N00014-75-C-0951-NR 049-308)National Science Foundation (Grant ENG71-02319-AO2)Jos6 M. Tribolet, Alan V. OppenheimHomomorphic filtering is a nonlinear signal-processing technique that has beenapplied to various deconvolution problems. The aim of this research is to study itsapplication to the "seismic deconvolution problem." Seismic data can often be modeledon a short-time basis as a convolution of a bandpass wavelet p(n) with a sequence ofimpulses r(n). Letting w(n) represent a short-time window, we may represent a seismictrace segment x(n) asx(n) = w(n)[p(n) * r(n)].The seismic deconvolution problem is to recover the sequence of impulses r[n] fromwhich the potential determination of the depths of the subsurface reflectors is possible.Short-time seismic data models exhibit characteristics that have to be accountedfor carefully in terms of their effects on homomorphic signal analysis. For example,their bandlimited nature has led to the generalization of homomorphic systems for thisclass of input signals.The short-time model has led to an understanding of windowing effects in the cep-stral domain, which, when conveniently explored, enable effective homomorphic waveletestimation by means of low-time cepstral gating. The corresponding estimates maythen be used to design optimum lag Wiener spiking filters that ultimately resolve thereflector series r[n]. This technique has been tested with good results on syntheticseismic data.4. ENHANCEMENT OF DEGRADED SPEECHU. S. Navy - Office of Naval Research (Contract N00014-75-C-0951-NR 049-308)Jae S. Lim, Alan V. OppenheimThe objective of this project is to develop speech enhancement techniques to increasethe intelligibility and quality of degraded speech when the degradation is caused by addi-tion of random noise. This research began last year and, as a starting point in ourresearch, we considered two existing speech-enhancement techniques, comb filteringand the INTEL (Intelligibility Enhancement by Liftering) system.The comb-filtering technique capitalizes on the periodic structure of speech wave-forms and attempts to eliminate the frequency bands where speech contributes littlePR No. 119(XVIII. DIGITAL SIGNAL PROCESSING)energy. By using this elimination process, we hope that more noise than speech willbe rejected. In our laboratory, some modification has been made on the adaptive comb-filtering method1 and the modified system was implemented on the PDP 11/50 computer.To determine the effect on the intelligibility score with the use of this system, nonsensesentences were processed and an intelligibility test is now in progress.The INTEL system is based on the concept that speech and noise contribute differ-ently to the autocorrelation function. In our laboratory, the original INTEL system 2was reduced to a simpler form, which resulted in two important advantages: a 40%reduction in computation time, and the conceptual simplicity of a less complex system.The system was implemented on the PDP 11/50 computer. As before, to determine theeffect of the system on the intelligibility score, we processed nonsense sentences andare testing their intelligibility.References1. R. H. Frazier, "An Adaptive Filtering Approach toward Speech Enhancement," S.M.and E.E. Thesis, Department of Electrical Engineering and Computer Science,M. I. T., June 1975.2. M. R. Weiss et al., "Study and Development of the INTEL Technique for ImprovingSpeech Intelligibility," Final Report No. NSC-Fi/4023, Nicolet Scientific Corpora-tion, Northvale, New Jersey, December 1974.5. ENHANCEMENT OF LOWPASS FILTERED SPEECHU.S. Navy -Office of Naval Research (Contract N00014-75-C-0951-NR 049-308)Elliot Singer, Alan V. OppenheimThis project is concerned with enhancing the quality of lowpass filtered speech byreinserting the missing spectral information. When only the low-frequency portion ofthe signal is present, it is possible that much of the missing high-frequency structurecan be determined from an examination of the available spectral energy and thus theoriginal speech can be reconstructed. This is especially true for voiced speech wherethe steady-state frequencies and amplitudes of the formants are well established.In a broad sense, an enhancement system for lowpass filtered speech would incor-porate algorithms for deducing the missing high-frequency structure and processingschemes for synthesizing the enhanced speech signal. Thus far, in our research wehave concentrated on techniques that keep the high-frequency characteristics fixed intime, and hence adaptive filters are not required.In order to achieve the most natural-sounding speech output, it was necessary tomake use of the available signal as directly as possible. This principle has been appliedto voice-excited vocoders with considerable success. In this system, the speechPR No. 119 100(XVIII. DIGITAL SIGNAL PROCESSING)synthesis is performed by extracting a subband of the original speech and processing itto make it a suitable excitation to the synthesizer. Linear prediction techniques may beused to flatten spectrally a lowpass speech signal whose spectrum has been broadenedthrough rectification. Theoretically, this approach should produce a signal whose spec-tral harmonics are closely related to those of natural speech.This technique was applied to an all-voiced sentence that had been lowpass filteredto 2 kHz. The resulting processed signal was superior in quality to the unenhanced orig-inal but suffered from a good deal of hoarseness. We believe that this hoarseness isattributable in part to the effects of spectral broadening and the manner in which thelinear prediction process operates on the speech signal.6. SPEECH ANALYSIS-SYNTHESIS BASED ON HOMOMORPHIC FILTERINGAND CCD TECHNOLOGYU. S. Navy - Office of Naval Research (Contract N00014-75-C-0951-NR 049-308)Thomas F. Quatieri, Jr., Alan V. OppenheimA nonreal-time speech analysis-synthesis system based on homomorphic filteringand the real cepstrum has been completed. We are developing a modification of thissimulation within the context of Charge Coupled Device (CCD) technology. In order toimprove this system, we are incorporating the complex cepstrum into the homomorphicalgorithm and studying the effects of phase on the quality of the synthesized speech. Oneimportant result is the sensitivity of the phase estimation to the time-domain windowduration, shape, and onset. This work should lead to a fixed phase compensation, whichcan be implemented by CCD technology.The FFT algorithm will be replaced by a skewed spectral analysis, the sliding chirp-z-transform, which is also well-suited to CCD technology. The sensitivity of this newspectral technique to the nonstationarity of the input speech waveform is being examined.We are also investigating novel methods of filtering the log spectrum in frequencyand the cepstrum in quefrency, which should reduce the raucous quality of vocoderspeech.7. SPEED TRANSFORMATIONS OF SPEECH SIGNALSU. S. Navy - Office of Naval Research (Contract N00014-75-C-0951-NR 049-308)Michael R. Portnoff, Alan V. OppenheimWe have designed and implemented on our PDP-11 computer a speech analysis-synthesis system based on the discrete short-time Fourier transform. This systemrepresents a speech signal by an appropriate set of time-variant parameters and hasPR No. 119 101(XVIII. DIGITAL SIGNAL PROCESSING)the property that when it is time-scaled and used in the synthesis procedure it producesspeed-transformed speech. We are now investigating two possible synthesis techniquesand beginning our investigation of feature-dependent speed transformations by selectivelytransforming only the stationary portions of the speech signal.PR No. 119 102

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