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We present a system for Internet communication that enhances traditional text-based chatting with real-time analysis and synthesis of the chat parties' facial expressions. It is composed of three main modules: a real-time facial expression analysis component, a 3D agent with facial expression synthesis and text-to-speech capabilities - a talking head, and a communication module. So far we have realized a prototype to find out attractive ways of communication on the Internet and are currently experimenting on how to utilize this new type of modalities in chat communication.

A novel Gaussian beam method, and a hybrid method, are developed for the efficient analysis of large reflector antenna systems, and large planar phased arrays, respectively. Several numerical results illustrating the speed and accuracy of these methods are described in this paper.

The author addresses the question of how channel errors affect the performance of subband coding systems. A basic subband coding model corrupted by either channel or quantization noise is considered. Power spectra of the output and reconstruction error are derived. Also, as an application of the above results, a method of finding optimum analysis/synthesis filter bank structures under minimum mean-square-error criteria is described. >

A unified analysis of monolithic microwave active inductors based on an amplifier-feedback topological description is introduced. Active inductors are classified based on the gain and feedback elements constituent to their design. Small signal, large signal and noise behavior are considered in the generic case. An illustrative example of the synthesis procedure for, as well as the measured results of a novel topology are presented.

The use of spectral analysis of heart sounds has been found to be an effective method for detecting different valvular disease and of studying the mechanism of heart action. The method used for spectral analysis is of crucial importance because the diagnostic ability of the technique depends on the accuracy of estimation of the heart sound spectrum. The authors present a new overdetermined forward-backward Prony's method (FBPM) which is superior to the modified backward Prony's method (MBPM) when applied to the analysis of first heart sounds (S1). The performance of the FBPM is compared with the MBPM using synthesised signals which are generated from a damped sinusoid model. Comparison of the results based on the least square error criterion show that the FBPM is more precise than MBPM with an improvement of up to 20%. The accuracy of this method makes it very attractive for diagnostic applications and investigating of the origins of S1.

For object-based analysis-synthesis coding (OBASC) of moving images, each moving object is described by three parameter sets defining its motion, shape and surface color. They are coded using an object-dependent parameter coding. Compared to block-based hybrid coding, OBASC requires the additional transmission of shape parameters. This is only justified, if it increases the overall coding efficiency. The efficiency of the block-based and the object-based coders depends to a large extend on the size of the moving objects. Objects smaller than 256 pels should be coded by a block-based coder whereas larger objects are preferably coded by an object-based coder. >

In this paper, we elaborate on the issue of analysis-synthesis consistency in sinusoidal coding. Our analysis is based on windowed sinusoids, and uses the same amplitude-complementary window as is used in the overlap-add synthesis. Reconstructions of the neighboring segments are taken into account when forming a particular analysis segment. Sinusoidal estimation is based on a perceptual criterion. In our new procedure, when analyzing the current segment we take advantage of the forward masking effect due to estimated sinusoids in the previous segments (possibly overlapping with the current segment). Experimental results verify that the number of sinusoids can be reduced significantly with our time masking model, without introducing perceptual artifacts in the reconstructed signal.

We compared six sound analysis/synthesis systems used for computer music. Each system analysed the same collection of twenty-seven varied input sounds, and output the results in Sound Description Interchange Format (SDIF). We describe each system individually then compare the systems in terms of availability, the sound model(s) they use, interpolation models, noise modelling, the mutability of various sound models, the parameters that must be set to perform analysis, and characteristic artefacts. Although we have not directly compared the analysis results among the different systems, our work has made such a comparison possible.

In this paper, a simple and efficient algorithm for designing digital sound watermark is presented. The main idea is by hiding the secret information in the amplitude of a certain frequency under 20Hz. This approach improved the sinusoidal analysis / synthesis model. Our approach can improve the sound watermark detection even after the signal has been compressed by MEPG lay-3. The experimental results illustrated the efficiency of our algorithm.

This article reviews Analysis, Synthesis, and Perception of Musical Sounds by James W. Beauchamp , 2007. 325 pp. Price: $89.95 (hardcover) ISBN-13: 978-0-387-32496-8

In this paper, different techniques for the creation of time coherent figurative animations from a video input are presented. The animations are generated using an analysis-synthesis approach. Information of the scene is extracted using only a 2D RGB image. No markers or depth planes are needed. After the analysis, the image is drawn again using the extracted information. To guarantee temporal coherence when redrawing the image, different alternatives have been explored: matching and interpolation on the parameters domain and gradient descent parameter update. All methods and variations are described and illustrated with images.

This paper presents an overview of the different works around gesture conducted at LIMSI Laboratory. Different kinds of capture devices are available, which allow us to deal with 2D or 3D gestures. Depending on projects, gestures are studied alone (monomodality), or combined with other modalities (speech). Different approaches are used, from analysis and synthesis to recognition and interpretation, along with knowledge representation.

Analytic measurement of visual parameters relevant to the labial production of speech as well as real-time, 3D-computer-animated models of the lips and of the face have been implemented on two coupled computers, so that synthetic lips alone or a whole facial model can mimic on line (or play back) the actual gestures of a natural speaker. The geometric measurements performed on the speaker’s lips and jaw are made through image processing of the front and profile view of the speaker’s face. Data are transmitted to a graphics computer through a control interface, which delivers the proper parameters to control the animation of the 3D models. The lip model uses five control parameters; the facial model uses one extra: jaw lowering. At present, the tongue is not controlled. We present the real-time techniques used for analysis, animation of the 3D models, and synchronization of the two processes. Finally, we evaluate the bimodal intelligibility of speech under five levels of acoustic degradation by added noise. We compare the intelligibility of the speech signal presented alone, with the lip model, with the facial model, and with the original speaker’s face. Our results confirm the importance of visual information in the perception of speech: the whole natural face restores two-thirds of the missing auditory intelligibility when the acoustic transmission is degraded or missing; the facial model (tongue movements excluded) restores half of it; and the lip model alone restores one-third of it.

This paper describes the implementation of a new parametric model of the glottal geometry aimed at improving male and female speech synthesis in the framework of articulatory analysis synthesis. The model represents glottal geometry in terms of inlet and outlet area waveforms and is controlled by parameters that are tightly coupled to physiology, such as vocal fold abduction. It is embedded in an articulatory analysis synthesis system (articulatory speech mimic). To introduce naturally occurring details in our synthetic glottal flow waveforms, we modelled two different kinds of leakage: a “linked leak” and a “parallel chink”. While the first is basically an incomplete glottal closure, the latter models a second glottal duct that is independent of the membranous (vibrating) part of the glottis. Characteristic for both types of leaks is that they increase de-flow and source/tract interaction. A linked leak, however, gives rise to a steeper roll-off of the entire glottal flow spectrum, whereas a parallel chink decreases the energy of the lower frequencies more than the higher frequencies. In fact, for a parallel chink, the slope at the higher freqencies is more or less the same as in the no-leakage case.

This is the final report summarizing the efforts performed under the grant "Analysis, Synthesis and Processing of Fractal Signals using Wavelets". This report includes a cummulative list of theses, conference presentations, and journal articles that have been supported either all or in part by the grant. Copies of the full texts have been sent to the program manager as they were written.

The chemistry of the key intermediate in the Maillard reaction, the Amadori rearrangements product, is reviewed covering the areas of synthesis, chromatographic analyses, chemical and spectroscopic methods of characterization, reactions, and kinetics. Synthetic strategies involving free and protected sugars are described in detail with specific synthetic procedures. GC- and HPLC-based separations of Amadori products are discussed in relation to the type of columns employed and methods of detection. Applications of infrared (IR) and nuclear magnetic resonance (NMR) spectroscopy for structural elucidation of Amadori products are also reviewed. In addition, mass spectrometry of free, protected, and protein-bound Amadori products under different ionization conditions are presented. The mechanism of acid/base catalyzed thermal degradation reactions of Amadori compounds, as well as their kinetics of formation, are critically evaluated.

A formal proof that bit allocation results hold for the entire class of paraunitary subband coders is presented. The problem of finding an optimal paraunitary subband coder, so as to maximize the coding gain of the system, is discussed. The bit allocation problem is analyzed for the case of the paraunitary tree-structured filter banks, such as those used for generating orthonormal wavelets. The even more general case of nonuniform filter banks is also considered. In all cases it is shown that under optimal bit allocation, the variances of the errors introduced by each of the quantizers have to be equal. Expressions for coding gains for these systems are derived. >

No reader of the Science of Logic interested in the history and philosophy of science, will fail to observe that in Hegel’s doctrine of method the two ancient ways of analysis and synthesis constitute the immediate precursor of the dialectic, which forms the crown of his whole system.

In the Karplus-Strong method, the synthetic sound is obtained by exciting a recursive comb-filter with a burst of noise. An analysis scheme is presented in this paper, which makes it possible to derive the optimal Karplus-Strong synthesis parameters from the recording of a real instrument. In addition, an excitation signal can be calculated and used in place of the usual burst of white or colored noise, in order to obtain more realistic syntheses.

A systematic approach to the description and analysis of robot's flexural characteristics is presented. Relations are derived for computing the total compliance matrix and deflection of the arm when considering the finite link stiffness, joint compliance and possible gear backlashes. The schema is applicable for serial, parallel or multi-arm structures. >

A subband coding system may be conveniently viewed as having two constituent components: the analysis/synthesis section pair and the coding section pair. The term analysis describes the process of splitting the input into critically-sampled frequency related subband signals while the term synthesis refers to the dual operation of interpolating and merging the signals to reconstruct the input. The coding section pair, which consists of an encoder and decoder, appears between the analysis and synthesis operations and enables the input to be represented at a reduced bit rate.

The authors present a solution to the problem of designing a class of analysis-synthesis digital filterbanks that exhibit the following properties: Given N center frequencies f/sub i/, the frequency response of the ith filter is unity at f/sub i/ and zero at the other frequencies, and the sum of the squares of the frequency responses is an allpass network. The design of typical filterbanks with linearly and logarithmically spaced frequencies is presented, and the frequency response is exhibited. These filterbanks are computationally efficient. >

For object—oriented analysis—synthesis coding an image analysis algorithm is required which automatically generates the parameter sets which describe moving 3D objects in an image sequence. Areas changed between two consecutive images are detected by means of change detecti()n. Special processing is carried out to compute areas which coincide with object boundaries and to eliminate areas which represent illumination changes. These areas are segmented into silhouettes of moving objects and uncovered background. The border of an ob ject silhouette is interpreted as the outermost contour of an object. These contours in combination with a simple function giving the z—distance between them provide a first estimiate for 3D shape of a model object. In order to improve the efficiency of motion analysis a concept for combining model objects with new parts of moving objects is proposed. Results of an automatic image analysis based on moving 3D objects are shown using video telephone test sequences.

This tried and tested book was first published by Stephen Rowland in 1986. Ernest Duebendorfer was involved in the second edition of 1994, and now Ilsa Schiefelbein has joined the team for the third edition. The philosophy and general content of this edition remains substantially the same as in previous versions, but the presentation has been thoroughly updated. All the figures have been redrafted, the tear-out maps and exercises are now gathered at the end …

Speech analysis‐synthesis in codecs and voice transformation often introduces distortion in the resynthesized output, particularly if the input is corrupted by noise. The harmonic plus noise model (HNM), particularly suited for voice transformation, divides speech signal into harmonic and noise parts. The harmonic part is synthesized by taking the summation of pitch harmonics up to a frequency called maximum voiced frequency, and the spectrum above this frequency is attributed to the noise part. Removal of nonharmonic components in harmonic part may result in noise rejection, while the errors in the estimation of harmonics may lead to distortion. Investigations were conducted to assess the speech enhancement obtainable through HNM on broadband noise corrupted speech signal. Listening tests showed that both the input and the resynthesized signals were perceived as distortion‐free for SNR greater than 10 dB. In the 0–10‐dB range, both the signals had no loss of intelligibility, but listeners rated the resynthesized signal as having higher quality. No enhancement was obtained for SNR values lower than 0 dB. Hence, for higher noise levels, spectral subtraction or other enhancement techniques should be employed along with analysis‐synthesis.

Analysis‐synthesis consists of constructing synthetic sounds from natural sounds by algorithmic synthesis methods. The models used for this purpose are of two kinds: physical models which take into account the physical characteristics of the instrument and signal models which take into account perceptual criteria. By combining physical and signal models hybrid models can be constructed taking advantage of the positive aspects of both methods. In this presentation I show how hybrid models can be adapted to specific instruments producing both sustained and plucked sounds. In these cases signal models are used to model the nonlinear source signal. The parameters of these models are obtained from perceptual criteria such as the spectral centroid or the tristimulus. The source signal is further injected into the physical model which consists of a digital wave guide model. The parameters of the physical model are extracted from the natural sound by analysis based on linear time‐frequency representations such as...

The author proposes an effective design structure for an analysis/synthesis filter bank with high computational efficiency. It can be shown that the resulting system filters are very efficient in terms of computational complexity and are relatively easy to design. Moreover, this design approach can be imposed on the system with relatively low reconstruction delays.

An impulse sequence of a pitch frequency is detected from a phase-equalized prediction residual of an input speech signal S(t), and a quasi-periodic impulse sequence is obtained by processing the impulse sequence so that a fluctuation in its pitch frequency is within an allowed limit range. The magnitudes of the quasi-­periodic impulse sequence are so determined as to minimize an error between the waveform of a synthesized speech obtainable by exciting an all-pole filter (18) with the quasi-periodic impulse sequence and the waveform of a phase-equalized speech obtainable by applying the input speech signal to a phase equalizing filter (5). Preferably, the quasi-periodic impulse sequence is supplied to the all-pole filter after being applied to a zero filter (10) in which it is given features of the prediction residual of the speech. Coefficients of the zero filter are also determined so that the error of the waveforms of the synthesized speech and the phase-equalized speech is minimum.

A linear predictive speech synthesis apparatus which receives a pitch parameter and linear predictive coefficients. The synthesis apparatus includes a device for producing an exciting source signal in response to the pitch parameter, and a device for filtering the exciting source signal in response to the linear predictive coefficients to produce a synthesized speech signal. A cascade frequency characteristic between spectral envelope frequency characteristics of the producing device and the filtering device is designated such as to correspond to a spectral envelope characteristic of an input speech signal.

Articulatory analysis/synthesis results using pitch‐synchronous analysis/synthesis were previously reported [Gupta and Schroeter, J. Acoust. Soc. Am. Suppl. 1 87, S39 (1990)]. In this paper, the interpolation characteristics of articulatory parameters within that framework for application to speech coding are investigated. One of the prime motivations for using an articulatory scheme for such an application is that articulatory parameters, polentially require a much slower rate of update. The parameters can be optimized once every few pitch periods and interpolated to obtain values at intermediate pitch periods. In this paper, two different interpolation schemes are presented and their results are compared. In the first scheme, the interpolation interval is fixed, say 100 ms. Since a pitch‐synchronous scheme is used, the nearest pitch marker is actually used. In the second method, the interpolation interval is made adaptive. The segmentation points within the speech utterance are determined using a smoothed spectral slope measure. The experiments indicate that the first scheme tends to smear out certain phonemes if the interpolation interval is selected to be too long. On the other hand, the second scheme requires automatic determination of appropriate interpolation intervals for different speech events.

The performance of the Burg method for speech analysis is compared to the autocorrelation and covariance methods. The criterion of goodness is the accuracy of the spectral approximation, filter stability, windowing requirements, data frame length, and spectral resolution. A mathematical comparison is presented for the simple first-order signal. Spectral comparisons are presented for a second-order speech-like signal. Real speech synthesis using the analysis results of the autocorrelation and Burg methods are subjectively compared. The results do not find any justification for preferring the computationally more complex Burg method.

The author favors a laissez-faire, "circus tent" approach to theory building and testing. Accordingly; he believes that analysis and synthesis are each of great importance in the development of theory -and would not suggest that scholars focus on one or the other. Unfortunately; there are serious problems in the research process in social psychology that hinder synthesis. These include a slavish fascination with "fashion, " which results in a tendency to neglect solid research and interesting theories simply because they are more than 10 years old. Unless this tendency is corrected, advances will continue to be disjointed. Reasons for this state of affairs are discussed.

This paper proposes the pitch synchro nous mel inverse LSP analysis-synthesis technique as a new coding method for LPC voiced residual. the pitch synchronous amplitude spectrum of the voiced residual has characteristic zeros. By applying the inverse LSP analysis in mel domain to this spectrum, the spectrum can be encoded with a high efficiency with order of analysis as small as 12. Then mel LSP inverse filter is proposed which can synthesize directly the voiced residual in the time domain from the mel LSP coefficients. It is shown that the voiced residual can easily be synthesized. For the voiced residual obtained from the forementioned analysis-synthesis system, the order of the model formel inverse LSP analysis, quantization characteristics and the interpolation performance were investigated, with the mean spectrum distortion of mel pitch synchronous amplitude spectrum as the evaluation measure. the optimization of each parameter was discussed. A pitch synchronous mel inverse LSP analysis-synthesis system was constructed based on the obtained result. A subjective quality evaluation was made for the synthesized speech, and it was verified that a high-quality speech can be synthesized at 4.8 kbit/s. These results indicate that the proposed method is applicable with merits to the low bit-rate vocoder.

In solving design problems designers use a combination of analytical serial thinking and holistic synthetic thinking. This can be seen both in how the process is described by design theorists and in the practical activities which designers undertake. In common with most other forms of design, the industrial design work which will be described is mainly concerned with defining the physical form of objects and this requires visual thinking. The whole process involves modelling the design ideas using various techniques, and for the visual thinking they are externalized in the form of drawings and three-dimensional representations. Using drawings is the quickest and most fluid modelling technique, and such drawings will often reveal the combination of analytical and holistic processes which the designer employs. Project work from Coventry's Industrial Design Transportation course is used to demonstrate such strategies.

On prepare une serie de derives du dimethyl-9,9 bicyclo [3.3.1] nonane substitues en -3 et/ou en -7. On etudie leur conformation par calcul de champ de force et spectrometrie RMN 13 C. Dans la plupart des composes, la presence des deux groupes methyle sur le pont impose la conformation chaise pour les deux cycles cyclohexane

Traditional speech analysis/synthesis techniques are designed to produce synthesized speech with a spectrum (or waveform) that is as close as possible to the original. It is suggested, instead, that representations of the synthetic and the original speech be matched at the auditory nerve level. This concept has been used in conjunction with the sinusoidal representation of speech analysis/synthesis suggested by McAulay and Quatieri [6]. Based on informal listening, the synthesized speech is natural, including some tonal artifact, and highly intelligible for various kinds of speech material, in both quiet and noisy environments. The inherent dominance property of the auditory nerve representation reduces the number of sinusoidal components needed for synthesis by approximately 70 percent, offering a potential for reduced data rate.

Many Pascal compilers are based upon a model which has two unpleasant properties: a large working set of procedures and lack of flexibility in object code structure. In this paper we analyse the source of these faults and describe an alternative model which avoids them. We then present the details of the key interface in the alternative model, as realized by a particular compiler.