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Publication numberUS3715512 A
Publication typeGrant
Publication dateFeb 6, 1973
Filing dateDec 20, 1971
Priority dateDec 20, 1971
Publication numberUS 3715512 A, US 3715512A, US-A-3715512, US3715512 A, US3715512A
InventorsJ Kelly
Original AssigneeBell Telephone Labor Inc
Export CitationBiBTeX, EndNote, RefMan
External Links: USPTO, USPTO Assignment, Espacenet
Adaptive predictive speech signal coding system
US 3715512 A
Abstract
In an adaptive, predictive coder for speech signals, the transmitted signal generally consists of an rms value, a pitch signal, a voice-unvoiced indication, and a number of parameter signals for adjusting the coefficients of a linear predictor. Transmission of these signals is improved in this invention by generating a low rate pulsive signal and by shaping its spectrum in accordance with the parameter signals. The pulsive signals thus act as a carrier for the parameters. The bandwidth required for transmitting the resulting composite signal, i.e., the modulated pulsive signal and the subsidiary signals, is substantially less than that of the original speech signal and somewhat less than that required for the transmission of predictively coded signals.
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Feb. 6, 1973 ADAPTIVE PREDICTIVE SPEECH SIGNAL CODING SYSTEM James Michael Kelly, Holmdel, NJ.

Bell Telephone Laboratories, Incorporated, Murray Hill, NJ.

Dec. 20, 1971 Inventor:

Assignee:

Filed:

Appl. No.:

References Cited UNITED STATES PATENTS 3/1964 Raisbeck ..l79/l5.55 R

12/1971 Atal ..179/l SA 11/1971 Atal ..l79/l SA OTHER PUBLICATIONS Atal & l-lanaver, Speech Analysis and Synthesis by Linear Prediction of the Speech Wave, JASA, 1971 pp. 637-655.

Primary Examiner-Ralph D. Blakeslee Assistant Examiner-Jon Bradford Leaheey Att0rneyR. J. Guenther et al.

[57] ABSTRACT In an adaptive, predictive coder for speech signals, the transmitted signal generally consists of an rms value, a pitch signal, a voice-unvoiced indication, and a number of parameter signals for adjusting the coefficients of a linear predictor. Transmission of these signals is improved in this invention by generating a low rate pulsive signal and by shaping its spectrum in accordance with the parameter signals. The pulsive signals thus act as a carrier for the parameters. The bandwidth required'for transmitting the resulting composite signal, i.e., the modulated pulsive signal and the subsidiary signals, is substantially less than that of the original speech signal and somewhat less than that required for the transmission of predictively coded signals. a a

8 Claims, 3 Drawing Figures l 7 l 22 4 COMBINER 25 silll fli iilt I 20 LlNEAR :B/M KHZ X: X] PREDICTOR PULSE I GENERATOR 0c I P TER I 23/ COM 0 J l RMS I +c W LPF 26 I DETECTOR T was PAR CLOCK COMPUTER Fawn,

J'F |4 |a I P|TCH8.VOlClNG SAMPLER DETECTOR v UV 1 H SI lg MULTIPLEX g ANALYZER- 51 d F|LTER SAMPLING RATE 52 2 25 D KHZ 2g 1 ll H3 2U i SPEECH i 55 E [;&-- SIGNAL l CODER 5N 8/M KHZ LINEAR VUV MULTIPLEX RMS SHEET 10F 2 FIG. I

COMBINER a PREDICTOR COMPUTER RMS DETECTOR COMPUTER DETECTOR PREDICTION PARAMETER PITCHEVOICING /IONI ANALYZER- SAMPLING RATE BKHz SPEECH SIGNAL CODER FIG. 2

COMBINER FILTER PATENTEDFEB s 1973 FROM AMP.------ ADAPTIVE PREDICTIVE SPEECH SIGNAL CODING SYSTEM This invention relates in general to the modification of signals to facilitate their transmission, and particularly to the reduction of their information rates and, hence, to the compression of their frequency bands. Its principal object is to compress the band of frequencies occupied by a speech message wave. A more general object is to apply new principles to the band compression or other modification of a speech message wave.

FIELD OF THE INVENTION The aim of efficient coding methods is to reduce the channel capacity required 'to transmit a signal with specified fidelity. To achieve this objective, it is often essential to reduce, so far as possible, redundancies in the transmitted signal. One well-known procedure for reducing redundancy is predictive coding. In predictive coding, that part of an applied signal that can be predicted from its past is subtracted from the signal to leave only an error, or difference signal, for transmission. For many signals, the spectrum of the difference signal is much flatter than the spectrum of the original signal. Thus, the difference signal is better suited to memoryless encoding than the original.

DESCRIPTION OF THE PRIOR ART Predictive coding of speech signals avoids many of the difficulties encountered with conventional vocoders and vocoder-like devices. Because of the rapidly changing structure of a speech signal, however, conventional linear prediction arrangements, often satisfactory for video coding, fail adequately to track a speech signal. This difficulty is overcome in an adaptive predictive coding arrangement for speech signals described in a copending application of B. S. Atal, Ser. No. 753,408, filed Aug. 19, 1968, now U.S. Pat. No. 3,631,520, granted Dec. 28, 1971. According to the Atal arrangement, the predictable portion of the speech signal is identified and, as in a conventional predictive coding arrangement, the remainder is treated as a difference signal for transmission. Because of the nonstationary nature of speech signals, however, the Atal system employs an adaptive predictor which is readjusted. periodically to match the time varying characteristics of the speech signal. Such an adaptive system has been shown to afford substantial transmission economies and to produce high quality synthesized speech.

Moreover, the parameter signals themselves have been found to represent adequately the spectral envelope shape of an input speech wave. Thus, in B. S. Atal U.S. Pat. No. 3,624,302, granted Nov. 30, 1971, a linear predictor vocoder system is described which is capable of producing high quality synthetic speech on the basis of predictor parameters and certain other speech signal values. A pitch detector is used, in accordance with the invention of that patent, as in a channel vocoder, to determine either the fundamental frequency of the voice wave or the absence of such a fundamental frequency. Speech signal parameters are continuously developed at a transmitter station, using the constraint that the applied speech wave at any instant is a weighted form of its past values. An Nth order linear predictor is adjusted to produce a minimum mean squared error signal. The predictor is basically a filter that has a spectral response that is a good match to the inverse of the spectral envelope of the input speech wave. The N predictor coefficients, together with the pitch frequency and the rms value of the input speech constitute the analyzer output signal, which may be transmitted to a receiver station. The receiver station employs a recursive filter in a feedback loop fed either by pulses at the pitch rate or by noise. The filter imposes a spectral envelope shape on the pitch pulses or noise to produce a signal which is a close match to the spectral envelope of the input speech. Amplifier gain in the system is adjusted to produce the correct rms value in the synthetic speech.

Such a predictive coding system permits efficient digital encoding of speech signals. Yet, it is often desirable to utilize analog facilities for the transmission of encoded speech, a task not readily possible with a predictive coding arrangement.

It is, therefore, an object of this invention, to achieve analog bandwidth compression and transmission economies in a linear prediction vocoder system.

SUMMARY OF THE INVENTION In accordance with this invention, a linear prediction vocoder analyzer is used to determine the spectral envelope of a speech signal, and an auxiliary spectral envelope signal is thereupon generated which is a frequency compressed value of the original spectral envelope. In essence, an analog signal is developed for transmission which contains all of the information needed to control a linear prediction vocoder synthesizer, for example, of the type described by Atal in U.S. Pat. No. 3,624,302, but which requires appreciably less bandwidth for transmission than either the original speech wave or prediction vocoder signals.

According to the invention, a coded signal for transmission is developed by generating a train of low rate pulsive signals and shaping the spectral envelope of the train in accordance with parameter signals which represent the spectral envelope of the speech signals. By employing a sampling rate in this spectral shaping process which is substantially lower than the rate of sampling of the input wave, the spectrum imposed on the low rate pulses is frequency compressed by a factor equal to the ratio of these two rates, The amplitudes of the pulses are thus varied to produce spectrally shaped pulses with the same rms values as the input speech. Preferably, the sampled data signal is converted into analog form prior to transmission. It may be combined with the pitch and voicing characteristic signals, for example, in a multiplexer, prior to transmission.

At the receiver, the inverse process is employed. Prediction parameters are recovered from the shaped pulsive signals and used, together with the pitch and voicing characteristic signals to synthesize a close replica of the original speech signal.

BRIEF DESCRIPTION OF THE DRAWING The invention will be fully apprehended from the following description of a preferred illustrative embodiment thereof, taken in connection with the appended drawing in which:

FIG. 1 is a block schematic diagram of an adaptive predictive speech signal coding arrangement which illustrates the principles of the invention;

FIG. 2 illustrates a linear predictor suitable for use in the practice of the invention; and

FIG. 3 is a block schematic drawing of a predictive speech signal decoder and synthesizer which may be used in the practice of the invention.

DETAILED DESCRIPTION OF THE INVENTION A transmitter station for developing narrow bandwidth signals for each of a number of voice channels is illustrated in the block diagram of FIG. 1. The station may include a number of individual speech signal codes through 10 each supplied with speech frequency signals, e.g., from transducers 11 through 11 and each delivering its coded output signal S to a transmission coder 12, of any desired construction. Coder 12 may be a multiplex unit, or the like, for combining the several supplied signals prior to transmission.

Each of coders 10 may be substantially identical in construction, with one 10,, being illustrated in detail. Speech signals from transducer 11 are passed through filter 13, which typicallyhas a cut-off frequency in the neighborhood of 4 kHz. The resultant signal is delivered to an analyzer which operates at a sampling rate of 8 kHz. Hence, the speech signal is sampled in unit 14 at a frequency of approximately 8 kHz under control of pulses from clock 15. Samples of the input signal are delivered to rms detector 16, predictor parameter computer 17, and voicing detector 18. In response, rms detector 16 develops a signal which characterizes the short-time energy of the signal, and voicing detector 18 develops two signals, namely, a signal P which denotes the pitch frequency or period of the input signal, and a voicing character signal V-UV which denotes whether the applied signal is voiced or unvoiced. Prediction parameter computer 17 develops a set of parameter values A=a,, a for each pitch period which implicitly describe the spectral envelope of the input speech signal below 4 kHz.

The construction, function, and mode of operation of the processing elements of the analyzer described hereinabove are disclosed in detail in the above-cited US Pat. No. 3,624,302 to Atal. As in the Atal system, the pitch frequency signal P and the voice-unvoiced signal V-UV are delivered to multiplexor 19 for eventual transmission. Unlike Atal, however, the rms signal and the several parameter signals are delivered, in accordance with this invention, to' a linear prediction synthesizer for further processing prior to transmission.

According, pulse generator 20 is .employed to develop pulses at a frequency proportional .to the sampling frequencies of the input signal. Suitable pulses are conveniently developed by dividing'pulses from clock by a factor C in divider network 21. For example, if samples of the input signal are developed at an 8 kHz rate, and pulse generator is adjusted to issue pulses at a 50 Hz rate, divider 21 is equipped with a divisor factor C=8 kHz/50 Hz=l60. The resulting low rate pulses from generator 20 are supplied to amplifier 22, whose gain is controlled by signals a from computer 23, to produce a signal at the output of combiner 24 whose rms value varies in accordance with the rms value of the input speech. The variable level pulses are added in combiner 24 to predicted values of the spectrally shaped pulses developed in predictor 25. The resulting signal, i.e., the sum of the input signal to combiner 24 and the predicted value, of it formed in predictor 25, constitutes the compressed bandwidth output signal.

Since the sampling rate used for spectral shaping is a factor M (analyzer sampling rate/synthesizer sampling rate) less than the sampling rate of applied signals, the spectrum imposed on the low rate pulses from generator 20 in the synthesizer is frequency compressed by a factor M over the inputspeech spectral envelope. For analog transmission, this signal is passed through a digital to analog converter, for example, low pass filter 26. Filter 26 typically exhibits a low frequency cut-off of 4/M kHz. The resulting analog signal, f,,, is delivered to multiplex unit 19 where it is combined with the pitch and voicing signals to produce a compressed bandwidth signal S for transmission.

Computer 23 may comprise an arithmetic processor using conventional analog circuit techniques or may comprise a special purpose computer or subroutine for evaluating certain arithmetic expressions. Its function is to control the amplitude of pulses produced at the lower rate which are used to synthesize an analog signal, narrower in bandwidth than the input signal, but containing all of the information needed for proper synthesis of the input signal. In essence, spectral shaping of pulses takes place in a linear predictor system (predictor 25 and combiner 24), which behaves as a linear system with a controlled impulse response. In this system, the input is known and the quiescent response is known. It is necessary, therefore, only to adjust the input to produce the desired output, a signal whose energy is equivalent to the mean-squared value of the input signal. The initial conditions of the linear system are taken into account by employing the principle of superposition, i.e., the output of the system is equal to the sum of the components due to the input signal and to the initial conditions of the system. Thus, if the output E of the system, due to the passage of an input signal pulse through it, is Z(t), and the output of the system, due to the initial conditions is Y(t), then where L represents the sample number at the beginning of a frame, e.g., a 20 ms interval, of K samples, and x, represents the compressed bandwidth signal. In the illustrated example, K C/M l/M. Signal X, is then represented by for i=L.

i=L+l, L+2, L+K.

where e,= a

e 0 for If Y, represents the response of the linear predictor to the last N output samples in the previous frame of K samples, at a time when there is a zero input to the predictor, the i" sample of the compressed signal is t t+ h (4) where Z, represents the response of the linear predictor to an impulse of unit magnitude and with an initial condition of zero. Therefore,

where i-J t-i I for i-j L.

Since Y,, Z, and E are known, the quadratic equation yields two values of a, of which the largest value is selected for use.

The computed value of a a for each set of parameter values A is thus supplied to control the gain of amplifier 22 and, hence, to control the magnitude of the pulse from generator 20 during the entire frame.

A suitable linear predictor, e.g., for use as element 25 in the system of FIG. 1, is illustrated in FIG. 2. It employs a transversal filter which includes a tapped delay line, a plurality of associated amplifiers and a summing network. Signal from combiner 24 are delivered to elements 27,, 27 27,, of the delay line and the intermediate delayed signals are passed through amplifiers 28 28,, 28 respectively. Amplifiers 28 are adjusted by parameter signals A from computer 17 and the resulting adjusted signals are summed in unit 29. The summed output signal represents a predicted value of the incoming signal based on a plurality of its past values. This signal is delivered to one input of combiner 24. A frame of past samples values X,, x are available at the intermediate points of delay line 27. These sample values are delivered to computer 23 FIG. 3 illustrates a receiver station, in accordance with the invention, which includes a number of separate speech signal decoders 30. Each decoder serves to analyze the supplied coded signal received from the transmitter and to develop therefrom the parameter values needed for synthesis. The composite signal input S is initially separated in transmission decoder 31 into the several channel signals S through S Each channel signal is then applied to the corresponding decoder wherein a replica of the input speech is developed and delivered, for example, to as associated loudspeaker 32.

One of the decoders, namely 30,, is shown in detail. Channel signal S is delivered to demultiplexer 33 to separate the several parameter values from one another. Pitch signal P is delivered to a synthesizer, operating at an 8 kHz rate, and is used to control the generation of periodic pulses in generator 34. The voicing characteristic signal V-UV is employed in the synthesizer to actuate selector 35. Pitch period pulses from generator 34 are amplified in controlled amplifier 36 in accordance with the value of control signal a, and supplied to one input of selector 35. This variable. amplitude pulse signal constitutes a voiced excitation signal for use in the synthesis operation. Similarly, un voiced excitation is produced in noise generator 37.

v Generator 37 typically produces a sequence of random pulses that are controlled in amplitude in amplifier 38 under control of a. Depending upon the voicing character of the speech signal, selector 35 delivers voiced energy from amplifier 36 or unvoiced energy from amplifier 38 to combining network 39.

The compressed bandwidth signal f developed at the transmitter station is supplied from demultiplexer 33 to an analyzer similar to the analyzer at the transmitter station. However, the receiver analyzer operates at a sampling rate of 8/M kHz. In the analyzer, signals from demultiplexor 33 are delivered to sampler 40 and the resulting samples f, are delivered to prediction parameter computer 41, and to rms detector 42. Computer 41, of essentially the same construction as computer 17 at the transmitter station, reestablishes parameter values A a a These values are delivered to a computer 43 and to linear predictor 44. Detector 42, essentially identical to detector 16 at the transmitter station, delivers the rms value of the compressed bandwidth signal to computer 43. Computer 43 is also supplied with an array X x,, x of samples of the past frame of signals from predictor 44. 7

As with computer 23 at the transmitter, computer 43 develops a control signal a for use in varying the amplitudes of both voiced and unvoiced synthesizer excitation signals, in amplifier 36 and 38, respectively. Computer 43 evaluates the same equation as a com puter 23 at the transmitter except that the value of K in Equation (2) is not constant at l60/M. Instead, it is equal to the pitch period P divided by MS or 1% kHz. Otherwise, computation with the supplied values is identical to that carried out at the transmitter.

Linear predictor 44 is supplied with output signals from combiner 39 and develops a signal proportional to the predictable portion of the output signal. Amplifier settings in predictor 44 are changed in accordance with parameter values A from computer 41 during both voiced and unvoiced speech intervals. Gate 48 is thus used to determine the precise instants in time at which the amplifier settings in predictor 44 are changed in accordance with parameter values A from computer 41. it allows the amplifier settings of predictor 44 to be updated to the current values of parameter values A of computer 41 whenever a pulse is present at the output of selector 46.

Selector 46, under control of the voiced-unvoiced signal, selects the output of 50 Hz clock 47 during unvoiced intervals, or selects the output of pulse generator 34 during voiced intervals. Since predictor parameter computer 41 determines a new set of parameter values at a 50 Hz rate (corresponding to the rate at which prediction parameters are changed in the synthesizer at the receiver), the amplifier settings of predictor 44 are effectively changed at the occurrence of only some pitch pulses. For example, if the pitch frequency is 100 Hz, then the amplifier settings of predictor 44 are changed for every other pitch pulse. Preferably, these amplifier settings are changed for every pitch pulse. This may be done conveniently by interpolating between consecutive sets of parameter values A. The interpolation arrangement described by Atal in his aforementioned patent may be used for this purpose.

The predictor output signal from unit 44 is returned to combiner 39 where it is added to the voiced or unvoiced excitation signal from selector 35 in conventional fashion. The output signal from the combiner is also delivered to a digital-to-analog converter; e.g., to lowpass filter 45, typically with a cut-off frequency of about 4 kHz. The resulting analog signal is delivered to loudspeaker 32,.

Speech signal decoder 30, thus serves essentially to generate parameter values for synthesis by means of an analysis process and, thereupon, to synthesize a signal on the basis of local excitation, a pitch signal, and supplied compressed bandwidth information.

What is claimed is:

l. A band compression system for the transmission of speech signals, which comprises:

a linear prediction vocoder analyzer supplied with a speech signal for developing a signal replica of the spectral envelope of said speech signal,

synthesizer means responsive to said spectral en velope signal for developing a frequency compressed spectral envelope signal,

means for transmitting said compressed envelope signal to a receiver, and

a linear prediction vocoder synthesizer at said receiver for developing a replica of said speech signal.

2. A band compression system for the transmission of speech signals, which comprises:

a linear prediction vocoder analyzer supplied with a speech signal for developing a signal representation of the spectral envelope of said speech signal,

means for generating a train of low rate pulsive signals,

means for shaping the spectral envelope of said pulsive signals inaccordarice with said spectral envelope signal,

means for transmitting said spectral envelope signal to a receiver,

means at said receiver for recovering said signal representation of the spectral envelope of said speech signal, and

means responsive to said recovered signal representation for synthesizing a replica of said speech signals.

3. A speech signal coder for developing reduced bandwidth signals for transmission, which comprises:

means for sampling an applied speech signal at a rate to preserve the characteristics of said speech signal,

means for analyzing said speech samples to develop parameter signals which represent the coefficients of a linear prediction of the present value of said speech signal on the basis of past values of said speech signal, the pitch, and the voicing character thereof,

means for generating a train of pulses at a rate lower than the rate of sampling of said speech signal,

a linear prediction synthesizer for shaping the spectrum of said pulses in accordance with said prediction coefficient values to produce a spectral envelope signal, and

means for combining said spectral envelope signal and said pitch and voicing character parameter signals for transmission to a synthesizer.

4. A speech signal coder as defined in claim 3,

wherein,

said train of pulses is generated at a rate commensurate with the rate at which said parameter values of said linear predictor are revalued.

5. A speech signal coder as defined in claim 3,

wherein,

the spectrum of said pulses is shaped in accordance with said prediction coefficient values to produce spectrally shaped pulses with the same rms value as said speech signal.

6. A speech signal coder as defined in claim 3,

wherein said synthesizer comprises:

means for recovering said coefficients from said transmitted spectral envelope signal and for recovering said parameter signals from said transmitted combined signal,

means for adjusting a linear predictor in accordance with said recovered coefficients,

means for predicting the present value of said received signal on the basis of said selected past values of it, and

means for developing a replica of said applied speech signal from said predicted values and from signals generated under control of said pitch and voicing character parameter signals.

7. In combination with the speech coder as defined in claim 3,

a speech decoder, which comprises,

means for recovering said spectral envelope signal, said pitch, and said voicing character parameter signals,

means for analyzing said spectral envelope signal at said lower rate to develop parameter signals which represent linear prediction coefficients and a signal representative of the rms value of said spectral envelope signal,

means responsive to said pitch signal for developing a train of pitch rate pulses,

means for generating a train of noise rate pulses,

means responsive to said parameter signals for predicting the present value of a reconstituted speech signal on the basis of past values of said reconstituted speech signal,

means responsive to said parameter signals, said predicted value signals, said rrns signal, and said reconstituted pitch signal for respectively shaping said train of pitch pulses and said train of noise pulses, and

means for selectively combining said shaped pulses and said reconstituted speech signal to produce a replica of said applied speech signal.

8. An adaptive predictive speech signal coding system, which comprises:

at a transmitter station;

means for sampling an applied speech frequency signal to develop samples at a first rate,

means for analyzing said samples to develop prediction parameter signals which characterize said sampled signal,

9 l synthesizer means operative at a second sampling means operative at said second sampling rate for rate responsive to said prediction parameter analyzing received analog replica signals to signals for developing an'analog signaling replica develop Predictivfi Parameter Signals which of said applied speech signal, characterize said received signal, and said second sampling rate being lower than said first Synthesizer means Operative at Said fi pl ng li rate, rate responsive to said prediction parameter signals for developing an analog signal replica of said applied speech signal.

means for transmitting said analog signal replica to a receiver station; and

at said receiver station; l0

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Non-Patent Citations
Reference
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Referenced by
Citing PatentFiling datePublication dateApplicantTitle
US3786188 *Dec 7, 1972Jan 15, 1974Bell Telephone Labor IncSynthesis of pure speech from a reverberant signal
US3903366 *Apr 23, 1974Sep 2, 1975Us NavyApplication of simultaneous voice/unvoice excitation in a channel vocoder
US3975587 *Sep 13, 1974Aug 17, 1976International Telephone And Telegraph CorporationDigital vocoder
US4022974 *Jun 3, 1976May 10, 1977Bell Telephone Laboratories, IncorporatedAdaptive linear prediction speech synthesizer
US4076960 *Oct 27, 1976Feb 28, 1978Texas Instruments IncorporatedCCD speech processor
US4130729 *Sep 19, 1977Dec 19, 1978Scitronix CorporationCompressed speech system
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US4852129 *Jul 21, 1986Jul 25, 1989Niravoice Inc.Data compression system using frequency band translation and intermediate sample extrapolation
US5522009 *Oct 7, 1992May 28, 1996Thomson-CsfQuantization process for a predictor filter for vocoder of very low bit rate
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US5737367 *Oct 7, 1994Apr 7, 1998U.S. Philips CorporationTransmission system with simplified source coding
US6370507Nov 28, 1997Apr 9, 2002Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung, E.V.Frequency-domain scalable coding without upsampling filters
US6732070 *Feb 16, 2000May 4, 2004Nokia Mobile Phones, Ltd.Wideband speech codec using a higher sampling rate in analysis and synthesis filtering than in excitation searching
DE3036440A1 *Sep 26, 1980Apr 16, 1981Hitachi LtdSprach-auswerter
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Classifications
U.S. Classification704/219, 704/208, 704/207
International ClassificationG10L19/04
Cooperative ClassificationG10L19/04, H05K999/99
European ClassificationG10L19/04