Contains research objectives and summary of research on seven research projects.U. S. Navy Office of Naval Research (Contract N00014-75-C-0951)National Science Foundation (Grant ENG71-02319-A02

Chan, David S. K.

Harris, David B.

Holt, Anthony P.

Kopec, Gary E.

McClellan, James H.

Mecklenbräuker, Wolfgang F. G.

Oppenheim, Alan V.

Portnoff, Michael R.

Quatieri, Thomas F., Jr.

Singer, Elliot

Tribolet, José M.

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XV. DIGITAL SIGNAL PROCESSINGAcademic and Research StaffProf. Alan V. OppenheimProf. James H. McClellanDr. Wolfgang F. G. Mecklenbr uker*Graduate StudentsDavid S. K. Chan Gary E. Kopec Thomas F. Quatieri, Jr.David B. Harris Michael R. Portnoff Elliot SingerAnthony P. Holt Jos6 M. TriboletI. SPEED TRANSFORMATIONS OF SPEECHU. S. Navy Office of Naval Research (Contract N00014-75-C-0951)Alan V. Oppenheim, Michael R. PortnoffThis research is directed toward transforming a speech signal into a new speechlikesignal that is perceived as identical to the original signal in all respects except that itis articulated at a rate different from that of the original speech. Such a technique isof interest in a variety of applications including learning a second language, developingtalking books for the blind, generation of stimuli for psychological experiments, enhance-ment of intelligibility, and interactively searching through large quantities of speechdata.Our current effort is directed toward three problems. The first is the formulationof an appropriate quasi-stationary representation for sampled speech signals. Thisrepresentation must have the property that time-scaling the parameters of the represen-tation corresponds to implementing the desired speed transformation. The second prob-lem is to develop a high-quality speech analysis-synthesis system based on thisrepresentation. As part of this effort, we have developed an efficient analysis-synthesissystem based on the discrete short-time Fourier transform. The third problem is todevelop a set of rules governing the local scale factor for the speed transformation. Spe-cifically, it is well known that a change in the rate at which speech is articulated doesnot correspond to a simple time scaling of the speech parameters independent of theuttered phoneme. We expect the scaling rules to take account of the fact, for example,that the formant trajectories of a vowel should be scaled, whereas the duration of a plo-sive burst should be left unchanged.2. ENHANCEMENT OF LOWPASS FILTERED SPEECHU. S. Navy Office of Naval Research (Contract N00014-75-C-0951)Alan V. Oppenheim, Elliot SingerThis research is part of a more general investigation concerned with enhancing thequality of degraded speech. When the degradation is caused by lowpass filtering, it isconceivable that the high-frequency components originally present could be determinedfrom an examination of the available low-frequency spectrum and the original speechthereby reconstructed. This is especially true for voiced speech where the overallVisiting Scientist, Philips Research Laboratories, Eindhoven, The Netherlands.PR No. 117(XV. DIGITAL SIGNAL PROCESSING)shapes of the spectral envelopes are well established. In a broad sense, therefore, anenhancement system for lowpass filtered speech would incorporate algorithms for gen-erating the missing high-frequency structure and processing schemes for synthesizingthe corresponding signals.In the initial work on this project we avoided the need for adaptive filtering by keepingthe characteristics of the high-frequency spectral shaping filter fixed in time. Also, theexcitation function used to drive this filter was derived from the lowpass filtered speechitself rather than through a pitch detection system. This technique, known as spectralflattening, seeks to smooth or flatten the spectral envelope while preserving the identityof the individual harmonic peaks.Some preliminary experiments with a speech sample lowpass-filtered to 2 kHz pro-duced a result that sounded somewhat unnatural. The system is now being improved bythe addition of more sophisticated spectral flattening techniques, as well as more care-ful structuring of the high-frequency synthesizer. Plans for the future call for theimplementation of an adaptive high-frequency filter and the possible extension of thehigh-frequency reinsertion technique to speech containing both voiced and unvoiced seg-ments.3. DESIGN AND IMPLEMENTATION OF VARIABLE CUTOFFDIGITAL FILTERSU. S. Navy Office of Naval Research (Contract N00014-75-C-0951)National Science Foundation (Grant ENG71-02319-A02)Anthony P. Holt, Wolfgang F. G. Mecklenbriuker, Alan V. OppenheimWe are constructing a digital filter with an easily variable cutoff frequency, usingthe technique of digital frequency warping. In brief, each unit delay of a digital filteris replaced by an all-pass filter. This permits the cutoff frequency of the digital filterto be varied over a wide range by modifying a single parameter. The particular filterthat is being constructed is lowpass in design with a cutoff frequency adjustable from1 kHz to 10 kHz. The cutoff frequency may be controlled from the front panel or by acomputer through an appropriate interface. To the user, the filter will behave like ananalog filter because of internal analog-to-digital and digital-to-analog conversion. Thevariable cutoff filter will be incorporated as a presampling and postsampling filter fordigital speech processing. With an easily controlled sharp cutoff filter of this kind, thesampling rate applied to the speech input and output in a speech-processing system caneasily be varied. The filter now being implemented in general has a nonlinear phaseeven when the prototype filter is linear phase. We have also developed a method forimplementing in a similar manner a variable cutoff digital filter that always maintainsexactly linear phase. While, at present, there are no plans for the actual constructionof the linear-phase filter, its application to speech-processing systems is being inves-tigate d.4. SPEECH ANALYSIS BY HOMOMORPHIC PREDICTIONU. S. Navy Office of Naval Research (Contract N00014-75-C-0951)Gary E. Kopec, Alan V. OppenheimLinear prediction is a generally accepted method for obtaining all-pole speech repre-sentations. In many situations (e. g. , nasalization studies) spectral zeros are importantand a more general modeling procedure is required. Unfortunately, the need for pitchPR No. 117(XV. DIGITAL SIGNAL PROCESSING)synchronization has limited the success of available techniques. In this project we havedeveloped and explored a new approach to pole-zero analysis, called homomorphicprediction, which seems to avoid the synchronization problem. A minimum-phase esti-mate of the vocal-tract impulse response is obtained by homomorphic filtering of thespeech waveform. Such a signal, by definition, has a known time registration. Linearprediction is applied to this waveform to identify its poles. The LPC "residual" (errorsignal) is computed by inverse filtering. This signal contains the information about thezeros. Its z-transform is then approximated by a polynomial either through a weightedleast-squares procedure (homomorphic prediction, using Shanks' method of findingzeros) or by spectral inversion followed by a second pass of LPC (homomorphic predic-tion involving "inverse LPC"). A preliminary evaluation on real and synthetic speechhas been carried out.In further work we intend to refine the general technique of homomorphic predictionand evaluate the pole-zero modeling within the context of a speech analysis-synthesissystem. Toward this end we intend to implement an analysis-synthesis system basedon homomorphic prediction and evaluate the potential improvement afforded by the inclu-sion of the zeros.5. APPLICATIONS OF HOMOMORPHIC FILTERING TO SEISMICDATA PROCESSINGU.S. Navy Office of Naval Research (Contract N00014-75-C-0951)National Science Foundation (Grant ENG71-02319-AO2)Alan V. Oppenheim, Jos6 M. TriboletHomomorphic filtering is a nonlinear technique that may be applied to deconvolutionproblems. Two homomorphic filtering approaches to deconvolution are available. Thefirst, homomorphic deconvolution, has been known for a while and has been used withsuccess in speech processing; here deconvolution is performed by linear filtering onthe cepstral domain. Another approach, called homomorphic prediction, has been devel-oped recently by us and performs deconvolution by linear filtering in the time domain,with a filter designed by combining homomorphic filtering with predictive filtering. Thistechnique has been applied with success to the compression of air gun signatures encoun-tered in marine seismic data. An exploratory use on lunar data has yielded promisingresults.The aim of this research project is to study the application of homomorphic decon-volution and prediction to the deconvolution of seismic data. Toward this goal a system-atic study of deconvolution of synthetic data will be made, to be followed by anevaluation of the performance of these techniques on real data.6. STRUCTURES FOR THE IMPLEMENTATION OF TWO-DIMENSIONALDIGITAL FILTERSNational Science Foundation (Grant ENG71-02319-A02)David S. K. Chan, James H. McClellanA new structure for efficient implementation of two-dimensional filters has beenderived from the recursion for Chebyshev polynomials. This new structure representsan improvement over the structures proposed previously by Mecklenbriuker andMersereau because the coefficients of the filter are well behaved. A comparative studyPR No. 117(XV. DIGITAL SIGNAL PROCESSING)of the two structures is under way in order to quantify these effects.Additional work is proceeding in an investigation of the computational structure forthese transformation filters. Such a study reveals the logical constraints on the orderof arithmetic operations and required storage organization for special-purpose hardwareimplementation.7. McCLELLAN TRANSFORMATIONS FOR TWO-DIMENSIONALDIGITAL FILTERSNational Science Foundation (Grant ENG71-02319-A02)David B. Harris, Wolfgang F. G. Mecklenbriuker, Thomas F. Quatieri, Jr.We have shown that a generalized form of the original McClellan transformation isa viable technique for the design of two-dimensional filters with quadrilateral symmetry.The essence of the technique involves mapping a one-dimensional finite impulse response(FIR) filter into a two-dimensional FIR filter through a substitution of variables. Severalstraightforward algorithms for estimating the transformation parameters have beendeveloped.We have also shown that two-dimensional filters designed by using a transformationpossess a series of highly efficient implementations that are more efficient than directimplementations and are even more efficient than a fast Fourier transform (FFT) imple-mentation for filters of order up to approximately 50 X 50. The implementation exploitsthe structure of the one-dimensional prototype filter. We have also explored techniquesfor overcoming potential difficulties inherent in these realizations, such as storage andeffects of finite register length.Finally, we have considered the limitations of McClellan transformations in pro-ducing optimum two-dimensional filters when the desired pass- and stop-band edges forideal two-dimensional filters coincide with contours of the transformations. Necessaryand sufficient conditions on the transformation guaranteeing that every optimum one-dimensional prototype maps to an optimum two-dimensional approximation have beenfound for the case of first-order transformations.PR No. 117

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