A metamodel is an important aspect of defining a modeling language. It specifies the language’s syntax through a set of constructs as well as how the language models are ought to be composed. Modeling languages, and thus their metamodels, are subject of constant evolution due to changing language requirements as consequence of business changes. Therefore, perceiving the essential aspects responsible for altering the structure of metamodels when a change requirement arises can become an issue. The Enterprise Ontology theory and its methodology (DEMO) provide ontological knowledge about organizations resulting in organizational self-awareness. Applying this methodology to the context of metamodeling can be a starting point for uncovering the essential aspects, i.e., the ontological knowledge regarding metamodel evolution. For that purpose, we modeled two diagrams using the DEMO methodology. The input for both diagrams was a set of coupled operations defined in Herrmannsdorfer’s evolutionary metamodeling research. In the end we stated the main conclusions of our work and themes for future work.

In order to continuously improve its knowledge and to identify problems and possible solutions, an organization requires understanding of the way business is aligned with the organizational strategy and how information systems are supporting the business. The paper presents a framework for describing and associating organizational concepts at multiple levels of detail using three separate areas of concerns: goals and strategy, business processes, and information systems. The framework is presented as an extension to the Unified Modeling Language (UML) using a standard UML Profile. The framework concepts are illustrated by modeling the purchase and sales business operations of a retail store from the strategic, process and information systems viewpoints.

A harmonic coder designed for operation at 4.8 kb/s is presented. Sinusoidal modeling of speech is reviewed and the use of harmonically related frequencies is extended to unvoiced and transition regions. The estimation of the fundamental frequency with emphasis on the fine tuning of the pitch estimates is discussed. The structure of the 4.8-kbit/s harmonic coder is described, and a detailed discussion of phase quantization issues is presented. Experimental results are presented which reveal a reverberant character in many speech utterances and difficulties in some classes of sounds (plosives, liquids, and voiced fricatives). These distortions are mostly produced by the quantization of phase, but they are also explained by the difficulties of the sinusoidal analysis/synthesis in transition regions when the frame length is large.>

A scheme is discussed for long-term prediction in CELP (code-excited linear predictive) coding using fractional delay prediction. This technique permits a more accurate representation of voiced speech and achieves an improvement of synthetic quality for female speakers. The higher complexity of this type of predictor relative to the classical one is its major disadvantage. Suboptimal schemes in which the search for the functional pitch delay is restricted to a neighborhood of an integer pitch estimate can be envisaged to decrease the computational load.>

The introduction of vector quantization in the harmonic coder framework can affect several sets of parameters, namely the amplitudes and the phases of the harmonic coefficients. A systematic experimental study is reported in which this type of quantization is compared with traditional scalar techniques for quantizing the harmonic parameters. The experiments showed that it is possible to quantize voice speech below 8 kb/s and still obtain high-quality synthetic signals.>

Two of the coding methods which have appeared recently, attempting to solve the problem of high quality speech coding at bit rates below 9.6 kbits/sec are harmonic and stochastic coding. The two represent distinct approaches to this problem and yield synthetic speech of a very different kind. In this paper, we discuss the advantages and disadvantages of both methods for processing voiced and unvoiced speech segments, and examine the potential benefits of combining the two methods in coding applications.

When trying to increase the efficiency of the multipulse speech representation through a reduction in the number of pulses per frame, the use of a pole-zero filter instead of the classical LPC filter appears to be an appropriate choice, for voiced segments. In this paper, the possibility and advantages of using this kind of filter are investigated. The filter design criterion is the minimization of the mean squared error (m.s.e.) between its output and the original voiced signal, when the excitation is a suitable periodic impulse train. This is, however, approximately equivalent to finding the filter that minimizes the m.s.e. between its transfer function and the signal harmonics, at the harmonic frequencies, taking into account both magnitude and phase. Some preliminary results are shown, and the conclusion is drawn that pole-zero filters actually are good candidates for a more efficient multipulse representation of voiced speech. The design of an algorithm for filter computation, and the incorporation of these ideas into the general multipulse framework are also discussed.

The area of Spectral Analysis has grown considerably in recent years. A multitude of methods are now available and yet no tools have been developed to guide the naïve user in the difficult process of choosing an adequate technique for his particular data problem. The purpose of this paper is to address, within a very general framework, the problem of choosing the "best" Spectral Analysis technique, from within a set of available methods and for a given class of data signals, according to some pre- -specified criteria. The foundations of the evaluation framework we propose are laid out in this paper. An in-depth study based on a particular evaluation methodology is reported, using a goodness of fit criterion derived from the works of Kedem and Slud [4,5]. Examples of this approach will be given for several data cases and several spectral analysis methods.

Multipulse LPC, as is often designated the model proposed by Atal and Remae, has been directed towards 9.6 kb/s speech coding. However, at such bit rate, the speech quality is not yet generally acceptable. This paper has a double purpose - one is to investigate the role of short-time phase in multipulse LPC. The other is to look for different modelling structures to be used with this method. This study was prompted by direct observation of the structure of the excitation, where patterns of pulses may be found which are associated with phase-correcting mechanisms of the LPC impulse response. Consequently, a new multipulse technique was developed, based on a special ARMA model formed by cascading an all-pole with an all-pass network. This new model will be referred to as the MAPAP (Multipulse All-Pole All-Pass) method. Another one was tried in which the synthetic speech is formed by combination of several MAPAP signals. We therefore denoted it "multichannel multipulse" method. The potential advantages of both single and multichannel models seem rather promising.

This paper deals with the problems associated with full-duplex scrambled speech communications over analog two-wire telephone networks. The goal is not to describe particular scrambling systems or methods, but to analyze the situation in which the two users speak at the same time and each should hear the other. Current telephone scrambling devices preclude this feature. A general half-duplex scrambling model is described as a base to the discussion of pseudo- and true full-duplex communications. Finally, a novel true full-duplex scrambling architecture based on a paper by Cox and Tribolet [11] is presented.

This paper addresses the ARMA spectral estimation problem, using as starting point, the Double Levinson Recursion Formulation proposed by Carayannis and al. in 1980 , to compute the AR parameters of an ARMA model. A theoretical analysis of the Double Levinson Recursion Formulation led to the derivation of a theorem from which the correct AR and MA orders may be computed. The algorithm is described in terms of an inner product formulation, using the theory of orthogonal polynomials. The Double Levinson Recursion Formulation is associated with a double lattice structure in a generalized Burg method fashion. When facing a purely AR process, the solution obtained is the same as that of Burg's MEM.

Spectral modeling is a key step in several speech processing applications. This paper presents a novel model capable of accurately handling the short-time spectrum of nonstationary voiced speech. As is well known, locally stationary, i.e., periodic, voiced speech exhibits a well-defined spectral line structure. It is shown in this paper that locally nonstationary voiced speech, i.e., speech with variations of both pitch and vocal tract, within the analysis window, still exhibits, in a generalized form, a spectral line structure, each line being formed by a series of well-defined components. The applications of this model to several speech processing problems, namely those involving speech prediction and pitch detection are presented, along with supporting experimental results.

Low bit-rate, good-quality speech coding is one of the fundamental goals of today's speech processing research. Present-day coding techniques, like APC and ATC, are able to achieve good-quality transmission only down to about 12 kb/s. Below this rate, their quality degrades rapidly. On the other hand, the various kinds of vocoders, which operate up to about 5 kb/s, have inherent quality limitations which cannot be overcome by an increase of the bit rate. In this paper, a new coding scheme is presented, which is based on a recently developed spectral model for nonstationary voiced speech, and it forms the basis of a waveform coder and a vocoder which are introduced in this paper, and which share the same basic structure. Experimental results are presented, which show that both systems yield significant bit-rate reductions relative to present-day schemes of equivalent quality.

Pole/zero modelling of speech has - been investigated for many years, yet the techniques developed so far have not proven satisfactory for most speech processing applications. Often these techniques separate the modelling problem in two: first and attempt is made to estimate the vocal tract impulse response; then an ARMA model is fit to such impulse response. The purpose of this paper is to discuss a different approach which is best-suited for speech analysis. This approach consist in first deriving from the speech signal a running approximation of a pseudo glottal excitation. Using a sequential minimization technique, the parameters of the ARMA model are then derived to fit a system whose input is the excitation signal and whose output is the speech signal.

Prediction plays a key role in many signal processing applications. Linear Prediction has, in particular, been extremely useful to the development of digital speech processing techniques and applications. There is however a growing need for improved forms of prediction. We discuss, in this paper, a form of non-linear prediction, namely, the prediction of the phase of speech signals. This study is conducted within a short-time analysis/synthesis framework and is based upon a new treatment of the classical speech production model. Experimental data are presented confirming the theoretical results. Finally the use of phase prediction to low-bit rate, high-quality coding applications is discussed.

The log likelihood measure has been widely used in speech recognition for comparing speech signals. Recently it has been proposed as a measure for assessing the quality of coded speech. In this paper we present an interpretation of the log likelihood ratio measure within the theoretical framework of a waveform coder distortion model.

The purpose of this paper is to describe a scheme for low-bit-rate transmission of speech. The scheme consists of a cascade of a codec with a post-processor. The codec is based on a modified version of the LPC Vocoder Driven Adaptive Transform Coding algorithm. The Post-Processor performs a short-time Fourier analysis/synthesis at the receiver output and, by exploring the known structure of the quantization noise introduced by the codec, it is capable of speech enhancement. The performance of this scheme will be demonstrated at 9.6 kb/s and at 7.2 kb/s.

Adaptive Transform coding techniques for speech communication have recently received considerable attention. The basic concept of these methods is to divide the speech into frequency components by a suitable transform and then encode them using adaptive PCM. Examples of designs and performance of adaptive transform coders will be presented for bit rates in the range of 8 to 16 kbs.

A wide variety of distance measures for comparing sets of LPC coefficients have been proposed. The most popular one has been the log likelihood ratio test as proposed by Itakura [1]. Although this measure is both computationally efficient, and theoretically sound, de Souza, in a recent paper [2], found that the measured distribution of LPC distances was significantly different from the one predicted by theory. De Souza then proposed alternative statistical tests for measuring LPC distances. In this paper we present both theoretical and experimental results which show that there is, in general, excellent agreement between theory and practice when using the log likelihood ratio to measure LPC distance. Included in the paper are discussions of the effects of LPC analysis method (covariance or auto-correlation), the use of fixed pre-emphasis, and the effects of additive, uncorrelated noise on the distributions.

This paper presents the results of a pilot study comparing four different speech waveform coding techniques of varying complexity. Coder transmission rates of 24, 16, and 9.6 Kb/s were used in the experiment. Subjective ratings and objective measurements of quality are obtained and compared. A number of conclusions are drawn concerning the quality and complexity, of different coding techniques. By comparing the objective measurements to the subjective ratings a number of conclusions are also drawn concerning the strengths and weaknesses of various (objective) quality measures of speech waveform coders.