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The areas of automatic speech recognition and speech synthesis are examined to ascertain what possibilities may exist for implementing them in Coast Guard Communication Stations. A discussion of the state of the art in both speech recognition and speech synthesis is presented. Concepts from the disciplines are given, as well as descriptions of many commercially available devices. We do not recommend that the Coast Guard pursue the development of a speech recognition system at this time. Several manufacturers are attempting to develop machines that will meet the Coast Guard's minimum requirements. Quite rapid progress is characteristic of the speech recognition field and suitable systems should be available in a few years. We do recommend that the Coast Guard research the implementation of speech synthesis technology at this time for the following: specific tasks, particularly weather reports, and general purpose use in Communication Stations.

Speech Analysis and Synthesis of the Punjabi Language

Bibliography: p. 406-426

Following an explanation and discussion of the importance of voice communications for military operations, including the environmental and propagation effects and ECM, the Lectures will outline: speech coding which is mainly concerned with man-to-man voice communication; speech synthesis which deals with machine-to-man communication; and speech recognition which is related to man-to-machine communication. All these are techniques which involve speech compression or speech coding at low-bit rates and are needed for transmitting speech messages with a high level of security and reliability over low data-rate channels and for other applications such as memory-efficient systems for voice storage and response. The themes above will be underpinned by a lecture on the nature of the speech signal (production, recognition, and perception) and complemented by other lectures on quality assessment of speech systems and standards which are crucial for the satisfactory deployment of speech systems. This Lecture Series, sponsored by the Avionics Panel of AGARD, has been implemented by the Consultant and Exchange Programme. Keywords: Voice communication; Military communication; Speech analysis; Human factor engineering; Speech recognition; NATO; AGARD; Man machine systems; Compandor transmission.

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Bibliography: p. 98-99

We give an overview of multilingual speech synthesis using the IPOX system. The first part discusses work in progress for various languages: Tashlhit Berber, Urdu and Dutch. The second part discusses a multilingual phonological grammar, which can be adapted to a particular language by setting parameters and adding language-specific details.

We give an overview of multilingual speech synthesis using the IPOX system. The first part discusses work in progress for various languages: Tashlhit Berber, Urdu and Dutch. The second part discusses a multilingual phonological grammar, which can be adapted to a particular language by setting parameters and adding language-specific details.

We give an overview of multilingual speech synthesis using the IPOX system. The first part discusses work in progress for various languages: Tashlhit Berber, Urdu and Dutch. The second part discusses a multilingual phonological grammar, which can be adapted to a particular language by setting parameters and adding language-specific details.

We give an overview of multilingual speech synthesis using the IPOX system. The first part discusses work in progress for various languages: Tashlhit Berber, Urdu and Dutch. The second part discusses a multilingual phonological grammar, which can be adapted to a particular language by setting parameters and adding language-specific details.

This report describes a new speech analysis/synthesis method. This new technique does not attempt to model the human speech production mechanism. Instead, we represent the speech waveform directly in terms of the speech waveform defined in a pitch period. A significant merit of this approach is the complete elimination of pitch interference because each pitch-synchronously segmented waveform does not include a waveform discontinuity. One application of this new speech analysis/synthesis method is the alteration of speech characteristics directly on raw speech. With the increased use of man-made speech in tactical voice message systems and virtual reality environments, such a speech generation tool is highly desirable. Another application is speech encoding operation at low data rates (2400 b/s or less). According to speech intelligibility tests, our new 2400 b/s encoder outperforms the current 2400-b/s LPC. This is also true in noisy environments. Because most tactical platforms are noisy (e.g., helicopter, high-performance aircraft, tank, destroyer), our 2400-b/s speech encoding technique will make tactical voice communication more effective; it will become an indispensable capability for future C4I.

A problem basic to the development of all digital telecommunication systems is the efficient digital encoding of speech signals for transmission. Many speech digitization algorithms have been proposed. This study, using as a research vehicle the homomorphic vocoder algorithm, is directed toward determining the time resolution and the frequency resolution required to faithfully reproduce a speech signal. The results reported are fundamental in that they are applicable to any speech digitization algorithm. The resolutions required for typical speaker situations are summarized.

Linear prediction is presented as a spectral modeling technique in which the signal spectrum is modeled by an all-pole spectrum. The method allows for arbitrary spectral shaping in the frequency domain, and for modeling of continuous as well as discrete spectra (such as filter bank spectra). In addition, using the method of selective linear prediction, all-pole modeling is applied to selected portions of the spectrum, with applications to speech recognition and speech compression. Linear prediction is compared with traditional analysis-by-synthesis techniques for spectral modeling. It is found that linear prediction offers computational advantages over analysis-by- synthesis, as well as better modeling properties if the variations of the signal spectrum from the desired spectral model are large. For relatively smooth spectra and for filter bank spectra, analysis-by-synthesis is judged to give better results. Finally, a suboptimal solution to the problem of all-zero modeling using linear prediction is given.

A MINIMUM-MEAN-SQUARED-ERROR TECHNIQUE FOR ESTIMATING THE COEFFICIENTS OF THIS FILTER FROM SPEECH DATA IS PRESENTED. This technique leads to a set of equations for the coefficient estimates which can be solved by a computationally efficient recursive technique known as Levinson's method. The filter derived by the above mentioned technique can be realized by any standard technique; however, a particularly interesting realization is in terms of a digital simulation of a non-uniform acoustic tube. It is shown that any stable all-pole filter can be realized as an acoustic tube and, moreover, that the Levinson recursion produces as a by-product exactly the reflection coefficients needed for such a realization. The report concludes by showing how the classical theory of orthogonal polynomials can be applied to the speech analysis/synthesis problem and used to derive many of the results obtained above by other means.

The note presents a tutorial survey of the mathematics that is used in the study of linear predictive filtering as applied to the analysis and synthesis of speech. Speech is modelled as the output of an all-pole filter that is driven by either a periodic pulse train or white noise. A minimum mean- squared-error technique for estimating the coefficients of this filter from speech data is presented. This technique leads to a set of equations for the coefficient estimates which can be solved by a computationally efficient recursive technique known as Levinson's method. The filter derived by the above mentioned technique can be realized by any standard technique; however, a particularly interesting realization is in terms of a digital simulation of a non-uniform acoustic tube. It is shown that any stable all-pole filter can be realized as an acoustic tube and, moreover, that the Levinson recursion produces as a by-product exactly the reflection coefficients needed for such a realization. The report concludes by showing how the classical theory of orthogonal polynomials can be applied to the speech analysis/synthesis problem and used to derive many of the results obtained above by other means.

Lt Stanford R. Wilde, Usn; A Thesis Written By Lt Wilde, "A Speech Analysis And Synthesis Scheme For Bandwidth Compression"

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