Electrical audio signal processing systems and devices – Monitoring/measuring of audio devices – Loudspeaker operation
Patent
1982-08-11
1985-09-17
Kemeny, E. S. Matt
Electrical audio signal processing systems and devices
Monitoring/measuring of audio devices
Loudspeaker operation
333118, G10L 100
Patent
active
045425248
DESCRIPTION:
BRIEF SUMMARY
BACKGROUND OF THE INVENTION
The present invention concerns a model of the acoustic sound channel associated with the human phonation system and/or music instruments and which has been realized by means of an electrical filter system.
Furthermore, the invention concerns new types of applications of models according to the invention, and a speech synthesizer applying models according to the invention.
The invention also concerns a filter circuit for the modelling of an acoustic sound channel.
In its most typical form, this invention is associated with speech synthesis and with the artificial producing of speech by electronic methods.
One object of the invention is to create a new model for modelling e.g. the acoustic characteristics of the human speech mechanism, or the producing of speech. Models produced by the method may also be used in speech recognition, in estimating the parameters of a genuine speech signal and in so-called Vocoder apparatus, in which speech messages are transferred with the aid of speech signal analysis and synthesis with a minor amount of information e.g. over a low information rate channel, at the same time endeavouring to maintain the highest possible level of speech quality and intelligibility.
Since the model of the invention is intended to be suitable for the modelling of events taking place in an acoustic tube in general, the invention is also applicable to electronic music synthesizers.
The methods of prior art serving the artificial producing of speech are divisible into two main groups. By the methods of the first group only such speech messages can be produced which have at some earlier time been analyzed, encoded and recorded from corresponding genuine speech productions. Best known among these procedures are PCM (Pulse Code Modulation), DPCM (Differential Pulse Code Modulation), DM (Delta Modulation) and ADPCM (Adaptive Differential Pulse Code Modulation). A feature common to these methods of prior art is that they are closely associated with signal theory and with the general signal processing methods worked out on its basis and therefore imply no detailed knowledge of the character or mode of generation of the speech signal.
The second group consists of those methods of prior art in which no genuine speech signal has been recorded, neither as such or in coded form, instead of which the speech is generated by the aid of apparatus modelling the functions of the human speech mechanism. First, from genuine speech are analyzed its recurrent and comparatively invariant elements, phonetic units or phonemes and variants thereof, or phoneme variants, in varying phonetic environments. In the speech synthesizing step, the electronic counterpart of the human speech system, which is referred to as a terminal analog, is so controlled that phonemes and combinations of phonemes equivalent to genuine speech can be formed. To date, these are the only methods by which it has been possible to produce synthetic speech from unrestricted text.
In the territory between the said two groups of methods of prior art is located Linear Predictive Coding, LPC, /1/ J. D. Markel, A. H. Gray Jr.: Linear Prediction of Speech, New York, Springer-Verlag 1976. Differing from other coding methods, this procedure necessitates utilization of a model of speech producing. The starting assumption in linear prediction is that the speech signal is produced by a linear system, to its input being supplied a regular succession of pulses for sonant and a random succession of pulses for unvoiced speech sounds. It is usual to employ as transfer function to be identified, an all-pole model (cf. cascade model). With the aid of speech signal analysis, estimates are calculable for the coefficients (a.sub.i) in the denominator polynomial of the transfer function. The higher the degree of this polynomial (which is also the degree of the prediction), the higher is the precision with which the speech signal can be provided with the aid of the coefficient a.sub.i.
The filter coefficients a.sub.i are however nonperspicuous from
REFERENCES:
J. Flanagan, Speech Analysis, Synthesis, Perception, McGraw-Hill, 2nd Ed., 1972, pp. 223-228.
Behaviour Research Method and Instrumentation, vol. 8, No. 2, Apr. 1976, (Austin, US), D. W. Massaro: "Real-Time Speech Synthesis", pp. 189-196, see in particular pp. 190, 191: The Synthesizer.
Journal of the Acoustical Society of America, vol. 61, Suppl. No. 1, Spring 1977, (New York, US), D. H. Klatt: "Cascade/Parallel Terminal Analog Speech Synthesizer and a Strategy for Consonant-Vowel Synthesis", p. S68, see abstract 114.
ICASSP 80, Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing, Apr. 9-11, 1980, Denver, IEEE (New York, US), vol. 3, J. L. Caldwell: "Programmable Synthesis Using a New Speech Microprocessor", pp. 868-871, see in particular Hardware Operation.
1971 IEEE International Convention Digent, published by The Institute of Electrical and Electronics Engineers, Inc., (New York, US), Y. Kato et al.: "A Terminal Analog Speech Synthesizer in a Small Computer", pp. 102, 103, see in particular figure 1.
Euroka Oy
Kemeny E. S. Matt
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