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US3631520A - Predictive coding of speech signals - Google Patents

Predictive coding of speech signals Download PDF

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Publication number
US3631520A
US3631520A US753408A US3631520DA US3631520A US 3631520 A US3631520 A US 3631520A US 753408 A US753408 A US 753408A US 3631520D A US3631520D A US 3631520DA US 3631520 A US3631520 A US 3631520A
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signal
signals
speech signal
speech
intervals
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Bishnu S Atal
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AT&T Corp
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Bell Telephone Laboratories Inc
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
    • H03M3/00—Conversion of analogue values to or from differential modulation
    • H03M3/02—Delta modulation, i.e. one-bit differential modulation
    • H03M3/022—Delta modulation, i.e. one-bit differential modulation with adaptable step size, e.g. adaptive delta modulation [ADM]

Definitions

  • an adaptive predictor is employed which is readjusted periodically to match the time-varying characteristics of a speech signal.
  • This invention relates to the efficient encoding of communication signals and to the reduction of the channel capacity required to transmit them. More particularly, it relates to the predictive coding of speech signals. It has for its object a reduction of redundancy in speech signals so that'the signals may more economically be transmitted to a receiver station.
  • 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 the redundancy of the transmitted signal.
  • One well-known procedure for reducing the signal redundancy is known as predictive coding.
  • predictive coding redundancy is reduced by subtracting from the signal that part which can be predicted from its past.
  • the first order entropy of the difference signal is much smaller than the first order entropy of the original signal; thus, the difference signal is better suited to encoding for transmission than the original signal.
  • Predictive coding thus offers a practical way of coding signals efficiently without requiring large capacity storage facilities.
  • One of the principal methods for efficiently encoding communication signals for transmission involves removing inherent signal redundancy through the use of prediction apparatus at both the transmitter and the receiver of a system.
  • the current value of the signal is estimated at both locations by linear prediction based on the previously transmitted signals.
  • the difference between this estimate and the true value of the signal is quantized, coded and transmitted to the receiver.
  • the decoded difference signal is added to the predicted signal to reproduce the input speech signal. So long as a good prediction of the present signal value can be made, efficient coding may take place.
  • speech is nonstationary so that a predictor with fixed coefficients fails to predict the value of a speech signal efficiently.
  • the speech signal is approximately periodic during voiced portions; thus, a good prediction of the present value can be based on the value of the signal exactly one period earlier.
  • the period varies with time so that the predictor must change with the changing period of the input speech signal.
  • speech signal statistics are not constant, it is necessary that the prediction parameters be varied in accordance with the nature of the incoming signal to adapt the predictor to the needs of the signal.
  • redundancy in speech signals is reduced by predicting the present value of the signal from its past and by subtracting the predicted value from the present value.
  • a form of predictive coding is employed in which the predictor adapts to changing signal conditions.
  • past signal intervals are selected for prediction that are comparable to individual pitch periods.
  • the extent of the period and the magnitude of the signal within the period are, in accordance with the invention, periodically redefined.
  • the parameter signals controlling the predictor are changed every milliseconds. Such an interval has been found to be satisfactory for accommodating the time-varying characteristics of the input speech signal.
  • the predictor parameter values are selected to minimize the power in the difference signal averaged over S-msec. intervals. As such, the predictor parameter values constitute slowly varying signals which can be transmitted efficiently.
  • the difference between the present value signal and the predicted value of the signal, if any, is eventually encoded and transmitted to a receiver station together with the slowly varying parameter signals which characterize the prediction.
  • the predictive coding system of this invention accurately reproduces a speech waveform rather than its spectrum. Listening tests show that there is only slight, often imperceptible, degradation in the quality of speech reproduced after transmission. In addition, experiments indicate that binary difference signals and predictor parameter signals prepared in accordance with the invention together can be transmitted at rates of less than 10 kilobits per second, or several times less than the rate required for ordinary pcm encoding with comparable speech quality.
  • the difference signal developed in accordance with the invention is the result of continuous efficient prediction, it has low or zero intelligibility. It may be used, therefore, together with the signals representative of the parameters of the adaptive predictor, which are themselves unintelligible, to provide secure transmission of speech signals. Only a recipient of the signals with the appropriate decoding apparatus can properly reconstruct the signals.
  • the predictor parameter signals may themselves be suitably scrambled for transmission. Only if the appropriate key to scrambling is known can they be properly recovered.
  • the difference signal and the parameter signal may be transmitted via independent channels as opposed to being multiplexed for transmission as a composite signal.
  • FIG. I is a block schematic diagram of a transmitter station which illustrates the principles of the invention.
  • FIG. 2 is a block schematic diagram of a receiver station constructed in accordance with the principles of the invention.
  • FIG. 3 is a block schematic diagram of an adaptive predictor suitable for use in the practice of the invention.
  • FIG. 4 is a block schematic diagram which illustrates the construction of a suitable predictor parameter computer
  • FIG. 5 is a block schematic diagram of a suitable arrangement for computing the values of parameter a used for adjusting an adaptive predictor used in the practice of the invention
  • FIG. 6 is a block schematic diagram of a transmitter station in accordance with the invention in which the difference signal and the several parameter signals are conveyed to a receiver station by separate transmission facilities, and
  • FIG. 7 is a block schematic diagram which illustrates the manner in which the parameter signals may be scrambled for transmission.
  • a predictive coding system for speech signals in accordance with the invention, includes: a transmitter (FIG. 1) for converting an input speech signal to a low-bit rate digital signal for transmission to a receiver; a predictor parameter computer (FIG. 4) to calculate the parameters of an adaptive predictor (FIG. 3); and a receiver (FIG. 2) to synthesize a speech signal from the received digital signal.
  • FIG. 1 A block diagram of a transmitter which illustrates the principles of the predictive coding system of the invention is shown in FIG. 1.
  • An input speech signal supplied at an input terminal is initially filtered in conventional low-pass filter l0 and sampled in sampling unit II.
  • the sampling rate is twice the cutoff frequency of the filter.
  • a suitable sampling rate for speech signals is 6 kilohertz, so that low-pass filter 10 has a cutoff frequency of 3 kHz.
  • Speech samples from sampler 11 are delayed by an interval of 60 samples msec.) by delay line 12 and delivered to one terminal of differencing network 13, for example, a subtracting network. (Since sampler 11 converts the input speech signal to a sequence of brief samples, i.e., to digital form, it is appropriate to consider the operation of the circuit on a sample-by-sample basis).
  • a predicted value 2,, of the speech sample, obtained by predieting (in network 30) the present value of the signal on the basis, for example, of the value of past samples r,,,, r is delivered to a second terminal of network 13.
  • the difference, 8 between 8,, and Z if any, issuing from the network is next supplied to an adjustable gain amplifier l4 and altered in amplitude by a factor Q.
  • the resultant signal is thereupon quantized to one of two levels, for example, in two-level quantizer 15.
  • the signal developed by quantizer is altered in amplitude by a factor HQ in amplifier l6 and supplied to one terminal of adder network 17.
  • the predicted value Z is supplied to the other terminal of adder 17.
  • predictor 30 uses only earlier samples (N-l, N2,...) of the reconstructed signal.
  • the current sample r of the reconstructed signal is formed after the difference signal 8 is quantized and added to the predicted value Z Adaptive predictor 30, which may be of the form illustrated in FIG. 3, is periodically adapted to accommodate changing signal conditions, for example, in accordance with parameter signals developed in computer 40. Details of a suitable computer are discussed hereinafter with reference to FIG. 4.
  • Predictor parameter computer 40 operates on signal samples supplied directly from sampler 11, and hence on signals in advance of their interaction in the differential operation, since signal S is delayed in unit 12 for a time sufficient to allow computer 40 to complete its operations.
  • the binary signal at the output of quantizer l5, parameter signals for adjusting the predictor, and the signal Q, representative of the gain of amplifiers 14 and 16, thus constitute components of the transmitted signal. They may be combined for transmission to a distant station in any desired manner.
  • the binary signal at the output of quantizer 15 may be supplied directly to multiplexer 18 and the parameter value signals b, K and or, and the signal 0 may also be supplied to multiplexer 18 for composite transmission to a receiver station.
  • the several signals may be transmitted via independent channels as shown in FIG. 6.
  • additional security may be achieved by scrambling the parameter signals according to a known code prior to transmission. A suitable arrangement is illustrated in H6. 7.
  • Scramblers suitable for cryptically encoding signals are known to those in the art. It is obvious that scrambled parameter signals and the difference signal may be transmitted to a distant station in any desired fashion, for example, by multiplexing as illustrated in FIG. 1 or by transmission over diverse paths as shown in FIG. 6.
  • Receiver 21 A block diagram illustrating the various functions performed by a receiver constructed in accordance with the invention is shown in FIG. 2.
  • Demultiplexer 21 serves to separate the various components of the composite signal received at an input terminal, namely, the quantized difference signal, signals denoting predictor parameters, and a signal representative of the gain of the amplifiers used at the transmitter.
  • the predictor parameters are supplied to adaptive predictor 30', which may be identical in all respects to adap tive predictor 30 at the transmitter.
  • the signal representative of the gain 0 is supplied to amplifier 22 and decoded difference signals are delivered to amplifier 22. After being adjusted in gain by the factor 1/0, the difference signal is added to a predicted value 2,, of the present value of the signal developed at the receiver, for example, in adder network 23.
  • the reconstructed samples n are delivered to adaptive predictor 30' and also supplied by way of low-pass filter 24 to an output terminal.
  • Low-pass filter 24 which has a cutoff frequency of one-half the sampling rate, smooths the supplied samples to produce an output speech signal r'(t). If there are no digital channel transmission errors, evidently predicted values Z are identical to values 2,, predicted at the transmitter, since predictor 30 is adjusted identically to its counterpart 30 at the transmitter. Hence, reconstructed sample r,,' is virtually identical to n, at the transmitter. It is apparent that the error between the reconstructed speech sample r,, and the input speech sample S is identical to the difference 5-8,.,- between the output of amplifier 16 and the input of amplifier 14.
  • Adaptive Predictor Two of the main causes of redundancy in speech are (l) quasi-periodicity during voiced segments and (2) lack of flatness of the short-time spectral envelopes.
  • redundancy due to the quasi-periodic nature of speech is reduced by a linear predictor consisting, for example, of a delay and a gain.
  • the z-transform of the predictor is given by where z represents a delay of K samples and b is an amplitude factor.
  • delay K corresponds nominally to a pitch period.
  • the factor b compensates for possible unequal amplitudes of the speech signal during contiguous pitch periods.
  • b is frequently greater than unity; the reverse is the case at the end of a voiced segment.
  • b is ordinarily close to zero.
  • Redundancy caused by the spectral envelope of speech is reduced, in accordance with the invention, by means of an eighth-order linear predictor.
  • the z-transform of such a predictor is given by
  • An eighth-order linear predictor has been found to substantially reduce redundancies due to three formants of the vocaltract transmission function and the spectral envelope of the vocal source.
  • FIG. 3 An adaptive predictor, which is suitable for speech signals, and is in accordance with these considerations, is illustrated in FIG. 3. It consists essentially of two separate linear predictor systems, which exhibit transfer characteristics in accordance with equations l and (2), and means for combining them.
  • Reconstructed signal samples, r, (delivered from adder network 17 of the transmitter, and correspondingly, from adder network 23 of the receiver, are delivered to storage unit 31.
  • This unit is equipped to store a variable digital signal y, for
  • n -l20, -1 l9,..., -l, 0, +l,..., +29.
  • storage unit 31 is actuated, for example, by a pulse from clock 37, such that the signal in storage location y replaces the signal stored at location y the signal at y replaces the signal at y and so on.
  • every 5 msec. a new group of samples is advanced into locations y yu to constitute a stored sequence of past samples. The locations, y y are vacated and made available to incoming reconstructed value signals r for the next 30 sample intervals.
  • l yn!( defines an input-output characteristic which corresponds to the form of equation (1 and specifies an output signal c for each supplied value of y,,.,..
  • the necessary factors, b and K, are supplied to arithmetic unit 32 from predictor parameters computer 40 (FIG. 1).
  • the resulting signals are delivered both to arithmetic unit 33 and to arithmetic unit 34.
  • Arithmetic unit 33 is programmed to develop values of u in accordance with the relation Fms The momentary value of r is supplied to arithmetic unit 33 from the input to adaptive predictor 30.
  • arithmetic unit 33 comprises a simple subtractor network.
  • Unit 35 may be a shift register or the like. It is reset every 5 msec. by a pulse, for example, from clock 37, to shift the signalsstored in locations 22 through +29 into the first eight locations and to free the locations through +29 for incoming signals.
  • the vacant storage locations are filled progressively with values of signal u, developed by arithmetic unit 33.
  • values of u, stored in unit 35 are delivered to arithmetic unit 36 which is arranged to compute values of c, in accordance with equation (2a), as follows:
  • Equation (2a) corresponds to the generalized relation of equation (2).
  • the neceswry amplitude factors a are supplied to unit 36 from predictor parameters computer 40 (FIG. 1).
  • Computed values of c; are delivered to arithmetic unit 34 wherein they are arithmetically added to values of c supplied by arithmetic unit 32 in accordance with equation (4) as follows:
  • arithmetic unit 34 may comprise an adder network.
  • the resulting valuesof Z constitute the predicted value of the incoming speech signal sample S and are delivered, as an output signal, to subtractor network 13 of the transmitter (FIG. 1) and, correspondingly, to adder network 23 of the receiver (FIG. 2).
  • the above-described arithmetic operations are carried on sequentially for each value of n from 0 through 29.
  • the integer N indicates the count of the current sample of the input signal, i.e., from sampler 12 (FIG. ll), minus 60 samples to take account of the msec. delay.
  • the integer N indicates a corresponding count within each unit.
  • Variables a and r are consecutively stored in storage units 3i and 35, respectively, in locations 0 through 29. Every 5 msec., both storage units are reset, as described above, and consecutive samples of r are again stored in locations 0 through 29 in storage unit 311, and consecutive samples u, are again stored in locations 0 through 29 and storage unit 35.
  • Predictor Parameters for the adaptive predictors at the transmitter and receiver stations are calculated in special computation apparatus which may be of the form illustrated in H6. 4. Such apparatus develops the predictor parameters necessary to ad just the predictor optimally despite the nonstationary, timevarying character of the input speech signals. Predictor parameters are recalculated every 5 msec. to ensure that prediction is efficient even when the speech characteristics are changing relatively fast.
  • Input speech samples S from sampler lll (FIG. I) are supplied to storage unit M which is equipped with sufficient storage capacity to accommodate an array w, in a configuration identical to that described above. Incoming samples are thus stored in the array as w w. w.,,..., w
  • the set of 30 newly installed samples constitutes a new frame of signals.
  • Arithmetic unit 42 includes individual computational units, 42a, 42b,..., 42m which operate in parallel to compute x, according to the equation for values of Fl5,..., 120.
  • a special purpose computer programmed according to the equation to be employed to evaluate these signal values or, alternatively, several individual arithmetic operations, e.g., multiplication, summation, rooting, and division, may be performed serially according to techniques well known in the an.
  • the computed array of values of x i.e., x are supplied in parallel to peak locating network 43 wherein the largest value of X is determined.
  • peak locating network 43 finds the value of j such that x, is the maximum of all values of Networks for picking the biggest from among a plurality of signals are well known in the art; a suitable one typically includes a progressively biased diode matrix.
  • the index of the largest selected value of X is designated K and is supplied as one parameter necessary to adjust the adaptive predictors at the transmitter and receiver locations.
  • n n nl n 0,...,29. (7) Values of signals in the array w, are supplied to arithmetic unit 45 from storage network 41.
  • the various computations outlined above are carried out serially in the order stated.
  • the suboperations e.g., the computation of values of x in arithmetic unit 42, b in computer 441, and u,, in arithmetic unit 45, are carried out in parallel circuits within those units.
  • Storage unit 46 thus stores an array of signal values u u v u,,,..., I129. Every 5 msec. the values u u are replaced by the valves u u The incoming samples are placed in the vacated storage locations u,,,..., [129. Thus, the signals u,,,..., 29, are consecutively stored as they are received in storage unit 46.
  • an array of signal values 11, are read out of storage unit 46 and transferred to arithmetic unit 47A.
  • This unit comprises 36 arithmetic units designated f f, f, f f; f f ,...,f which operate in parallel.
  • Each unit serves to compute one value offaccording to equation (8) as follows:
  • Arithmetic unit 473 preferably comprises an array of eight individual units operating in parallel to evaluate the several values of g.
  • the resultant array, g,,..., g,,, designated G, is delivered every msec. to computer 48.
  • Array :1 is also delivered to 0 computer 49.
  • Computer 49 constitutes an arithmetic unit arranged to evaluate values of Q according to the relation Arithmetic units for obtaining products, summations, differentials, absolute values and so on, are well known to those skilled in the art. Values of Q thus evaluated are used both at the transmitter and at the receiver to set the gains of the several adjustable gain amplifiers used in the predictive networks. At the transmitter, values of signal Qare used to set the gains of amplifiers l4 and 16; at the receiver to set the gain of amplifier 22.
  • the various predictor parameters and gain factor Q are recalculated every 5 msec. These calculated values are held fixed for a duration of 5 msec., the period over which the predictor parameters have been optimized. Due to the 10 msec. delay of incoming signals at the transmitter, the predictor parameters computer calculates the parameters ahead of the time they are needed at the transmitter. The adaptive predictors are reset just before the arrival of the first speech sample of each frame at the transmitter.
  • the array of signals F representative of values off developed in arithmetic unit 47A, are supplied, respectively, to arithmetic units 51.
  • Arithmetic unit 510 for example, develops a value of h, in accordance with equation 12), as follows:
  • arithmetic unit 510 comprises a square rooting device. Values ofh h are evaluated in arithmetic unit 51b in accordance with the relation shown in equation (l3), viz, lHIJ/ LlsF v-w
  • arithmetic unit 51b comprises a plurality of individual units for developing a quotient signal. The necessary value of h, is delivered to arithmetic unit 51!) from storage unit 52.
  • the magnitude of e is selected in accordance with the relative signal magnitudes accommodated by units 51, to be insignificant as far as signal evaluation is concerned, but sufficient to avoid the divide-by-zero ambiguity. If desired, switch 55 may be used to open the e circuit except when zero signal is detected.
  • the apparatus described herein represents merely one suitable manner of carrying out the necessary operations to adaptively predict the values of a speech signal to promote efficient coding for transmission.
  • Numerous alternative techniques may be employed for the evaluation; in fact, many of the operations may be programmed for evaluation by a special purpose computer.
  • the signals prepared for transmission may be combined in any desired fashion or, in the alternative, may be transmitted separately to achieve secure transmission of the speech signals.
  • the quantizing noise appearing at the output of the receiver is essentially white in nature (flat spectrum). Frequently it is desirable that quantizing noise have a nonflat spectrum. For example, noise whose spectrum is weighted down at high frequencies may be subjectively less annoying. Any desired noise spectral characteristics can be obtained by employing a suitable preemphasis network before low-pass filter 10 in the transmitter and a dcemphasis network after low-pass filter 24 at the receiver.
  • a suitable preemphasis characteristic for speech signals is one which is flat up to about 500 Hz. and rises at about l db. per octave between 500 and 300 Hz.
  • lt is not necessary that a preemphasis network be used prior to lowvpass filtering. It may, for example, be used just after the sampler 11. Similarly, the dcemphasis network may be used just prior to low-pass filter 24 in the receiver.
  • Speech signal processing apparatus which comprises:
  • said characteristics of said speech signal represented by said parameter signals include the extent of selected past pitch periods and the magnitude of signals within said pitch periods.
  • Speech signal processing apparatus as defined in claim 1, wherein new parameter signals are developed every 5 milliseconds.
  • said means for predicting the present value of said applied speech signal comprises,
  • a linear predictor characterized by a z-transform given by where b is a factor representative of signal values during consecutive selected signal intervals, K is a number representative of the duration of consecutive pitch periods of said applied signal, a, are amplitude factors representative of the short time spectral envelope of said speech signal, and N represents a selected number of said factors a,,,.
  • a communication system for conveying the information content of a speech signal over a channel of relatively small capacity which comprises, in combination:
  • Apparatus for predicting the present value of a speech plitudes of correlated signals in a number of selected coni l f i past which comprises; secutive mtervalsof said applied speech signal;
  • Apparatus for developing parameter signals for use in the predictive coding of speech signals which comprises, in combination:

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GB1266929A (xx) 1972-03-15
ES371061A1 (es) 1971-08-01
JPS5021203B1 (xx) 1975-07-21
SE351066B (xx) 1972-11-13
NL176822C (nl) 1985-06-03
NL176822B (nl) 1985-01-02
FR2016969A1 (xx) 1970-05-15
DE1941336C3 (de) 1979-12-13
NL6912534A (xx) 1970-02-23
DE1941336B2 (de) 1979-04-19
DE1941336A1 (de) 1970-02-26
BE737031A (xx) 1970-01-16

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