CA1224876A

CA1224876A - Data compression system and method for audio signals

Info

Publication number: CA1224876A
Application number: CA000455535A
Authority: CA
Inventors: Charles S. Weaver
Original assignee: SRI International Inc
Current assignee: SRI International Inc
Priority date: 1983-12-12
Filing date: 1984-05-31
Publication date: 1987-07-28
Also published as: DE3490580T; JPS61500998A; WO1985002529A1; NL8420060A; GB2165426B; GB8519232D0; DE3490580C2; GB2165426A

Abstract

ABSTRACT OF THE DISCLOSURE
A data reduction system is disclosed which includes an analog to digital converter for converting an analog signal to digital sample signal form, a digital compression filter for compression filtering the digital sample signals, and an encoder for truncated Huffman encoding the compression filter output. A decoder for decoding the encoded signal, a digital reconstruction filter for decompression filtering of the decoded signal, and digital to analog converter means are included to reconstruct the analog signal. The digital compression filter has a transfer function which includes zeros on the unit circle in the Z-plane at substantially zero degrees from the origin, while the digital reconstruction filter has poles on or inside the unit circle in the Z-plane at substantially zero degrees from the origin. The transfer function of the digital compression may also include zeros on the unit circle in the Z-plane at at least one of the following pairs of angles, +41.41, +60°, +90°, +120° and +180°, in which case the transfer function of the digital reconstruction filter includes poles on or inside the unit circle at corresponding angular positions. The poles of the reconstruction filter may be momentarily movable inwardly inside the unit circle in response to a detected error signal to speed recovery from such error signal.

Description

12~4876 DESCRIPTION

DATA COMPRESSION SYSTEM AND
METHOD FOR AUDIO SIGNALS

BACKGROUND OF THE INVENTION

Systems which include means for converting analog signals to digital form, then compression filtering and Huffman encoding the signals for recording or for transmission to a remote location, together with playback or receiver means which include a Huffman decoder, a digital reconstruction filter and means for converting the decoded and filtered digital signals back to analog form are shown in U.S. Patent No. 4,449,536 issued May 22, 1984, and in an article by U.E. Ruttimann and H.V. Pipberger entitled "Compression of the ECG by Predlction or Interpolation and Entropy Encoding", IEE Transactions on Biomedical Engineering, Vol. BME-26, No. 11, pp. 613-623, Nov. 1979. A similar system is shown in an article by K.L. Ripley and J.R. Cox, Jr. entitled, "A computer System for Capturing Transient Electrocardiographlc Data", Pro.
Comput. Cardiol. pp. 439-445, 1976. Wi-th the present invention, the average bit rate of an ..

analog-to-digital converted audio signal, such as a music, electrocardiogram, or electroencephalogram signal, is reduced sufficiently to allow for digital transmission thereof over low-grade transmission lines and/or recording and playback of a worthwhile quantity of signal using a relatively small amount of recording medium and employing known digital recording and playback techniques.

SUMMARY OF THE INVENTION

Audio signals, such as music, to be transmitted, or recorded, are converted to digital form by analog to digital converter means. The digital signals then are supplied to a digital compression filter to generate digital, compressed signals. The compressed audio signals are supplied to an encoder, such as a truncated Huffman encoder, for encoding the same. The digital output from the encoder is recorded by use of digital recording means, and~or transmitted to a remote receiving loca~ion. At a playback unit or receiving station the encoded signal is decoded by a decoder, and the decoded signal is supplied to a digital decompression filter. The output from the decompression filter is converted to analog form by digital to analog converter means to provide a reproduction of the audio signals. A digital compression-decompression filter combination is used which minimizes the average bit length of the recorded, or transmitted, digital signal words. The transfer function of the digital compression filter means includes zeros on the unit circle in the Z-plane, and the transfer function of the digital decompression filter includes poles on or inside the unit circle in the Z-plane at the same angular positions as the zeros of the compression filter. Zeros and poles of the compression and reconstruction filters, respectively, are limited to angular positions of 0, +41.41, +60, +90, ~120, and 180. Compression filter operation is performed without truncation, or round-off,whereas decompression filter operation is with truncation, or round-off. The resultant system frequency response for analog signals is high-pass which may include one or more small high frequency notches, For music signals, the resultant filter has a low frequency cut-off of between 0-15 Hz to accommodate audio frequency signals which range, approximately, from 15 Hz to 20,000 Hz.
An unstable compression-decompression filter combination results when the decompression filter transfer function includes poles on the unit circle of the Z-plane. For such arrangements, transfer of the output from the Huffman encoder may include use of an error-checking code and error detecting means for detection of errors in said transfer to the Huffman decoder An error signal is produced in response to the detection of an error in such digital signal transfer, which error signal is supplied to the decompression filter for use in momentarily mov:Lng the poles of said filter inwardly, inside the unit circle thereby enabling the system to recover from said signal errors.
The above-described error signal detection and inward movement of the poles of the transfer ;

1 ~24~i7~

function in the 7,-plane of the decompression filter in response to error detection also may be used in those systems having stable compression-decompression filter combinations to accelerate recovery from signal errors.
Instead of using an error checking code and error signal detecting means in those systems which include an unstable compression-decompression filter combination, the system may be operated to periodically transfer a series of actual signal values from the A/D converter to the reconstruction filter thereby periodically "reinitializing" the reconstruction filter when errors have occurred.
In the case of music signal compression, transmission of actual signal val~es every~ say, 6 to 16 milliseconds would be adequate. The number of successive actual signal values reguired to be periodically transmitted is dependent upon the order of the decompression filter; the number of actual signal values sent being equal to said order.

BRIEF DESCRIPTION _ THE DRAWINGS

The invention will be better understood from the following description when considered with the accompany:ing drawings. ,[n the drawings, wherein like reference characters refer to the same parts in the several views:
Figs. lA and lB together show a block diagram of a data reduction system; a digital recording and transmitter section being shown in Fig, lA and a playback and receiver section being shown in Fig.
lB;

1~4876 Fig. 2 shows a waveform and graphic representations of signals appearing at various locations in the data compression system shown in Figs. lA and lB;
5Fig. 2A shows the frequency response of high frequency deemphasis and high frequency emphasis filters which are included adjacent the input and output, respectively, of the data reduction system;
Fig. 3 is a graphic representation of encoded difference signals showing the format employed for encoding those difference signals which are outside a predetermined signal range;
Fig. 4 is a graph for use in showing the relationship between the probability that a digital sample signal value will occur within a certain quantization level and the size of the quantization level;
Fig. 5 shows zeros of a second order compression filter transfer function on a unit circle in the Z-plane;
Fig. 6 shows a plurality of z-transform zero positions which may be employed in the compression filter embodying this invention;
Fig. 7 is a graph showing the frequency response of three different compression filters having ~eros on the unit circle of the z-transform at some of the positions identified in Fig. 6;
Fig. 8 is a block diagram showing details of a compression filter of the type which may be used in the present system;
Fig. 9 is a table showing a truncated lluffman code of a type which may be used in the present invention;
Fig. 10 shows the zero-pole pattern of a compression-reconstruction filter con,bination which 1~24876 may be employed in the present system which results in a stable system without the need for error detection;
Fig. 11 shows the frequency response of the compression-reconstruction filter combination having the ~ero-pole pattern illustrated in Fig.
10;
Figs. 12A and 12B are similar to Figs. lA and lB, respectively, but showing a system which includes a check bit generator and error checking means for use in momentarily moving the poles of the reconstruction filter inwardly when a bit error is detected;
Fig. 13 shows a zero-pole pattern of a compression-reconstruction filter combination in which poles of the reconstruction filter are momentarily moved inwardly to accelerate recovery from transients;
Fig. 14 is a block diagram showing details of a reconstruction filter which may be used in this invention;
Fig. 15 shows the zero-pole pattern of another compression-reconstruction filter combination which may be used in systems embodying the present invention; and Fig. 16 is a graphic representation of Huffman encoded signals present in a system wherein actual digital signal values are periodically transferred to the reconstruction filter to periodica:Lly reinitialize the same.
Reference first is made to Fig. lA
wherein the digital recording and transmitting portion of a data compression system is shown comprising an analog to digital converter (A/D
converter) 20 for conversion of an analog audio signal f(t) into digital form, the nth sample from the analog to digital converter 20 being identified as fn. At A of Fig. 2, an analog signal 22 is shown which comprises an input to the analog to digital converter 20. For purposes of illustration, the audio input signal may comprise a music signal which ranges in frequency from approximately 15 to 20,000 Hz. The form of the analog to digital converter output, shown at B of Fig. 2, comprises samples fn 1 through fn+i of equal length words. The analog to digital converter 20 operates at a sampling rate established by control signals from a timing and control unit 24 supplied thereto over timing line 26.
As employed herein, line 26 from the timing and control unit 24 represents a plurality of timing circuit outputs, one or more of which are supplied to the system elements for proper system timing and control. Inputs also are supplied to the timing and control unit over line 28 for control thereof by signals from various other system elements. The A/D converter 20 operates in a conventional manner at a fixed sampling rate and with a fixed word length output. For purposes of description only, and not by way of limitation, the A/D converter may operate at a sampling rate of 44KHz and with a 14 bit word length.
The output from the A/D converter 20 is supplied to a linear digital compression filter 30 through a digital filter 23 which deemphasizes the high frequency portion of the digital audio frequency signal from the A/D converter 20 to reduce the signal entropy. The frequency response of filter 23, together with the frequency response of a digital filter 75 included in the playback and 12~48~, receiver portion of the system is shown in Fig. 2A.
For simplicity, the digital output from filter 23, as well as the digital input, is identified as fn.
Obviously, an analog filter having a similar S frequency response may be included at the input to the A/D converter 20 in place of the digital filter 23 at the output therefrom.
For present purposes, the digital compression filter 30 is shown to include an estimator 32 and subtracting means 34. The estimator 32 provides an estimate of fn here identified as fn based upon actual samples occurring both before and after the sample fn to be estimated. Estimators for providing such estimated fn values are, of course, well known. A difference signal ~ n is produced by the compression filter 30 comprising the difference between the actual signal input fn and the estimated signal value fn by subtraction of the estimated value from the actual value at subtracting means 34, as follows:

n = fn ~ fn (1) In Lhe graphic signal representation of the compression filter output shown at C in Fig. 2, difference signals a n~ ~ n+1~ ~ n+2~-----~n+i are shown. In accordance with one feature ofthis invention, arithmetic operations of the digital compression filter 30 are performed without truncation or round-off whereas arithmetic operations of an associated digital decompression, or reconstruction filter, described below, are performed with truncation, or round-off. As seen in Fig. 2C, the compression filter output comprises 12~487fi untruncated compressed signals which are18 bits in length.
It here will be understood that the present invention is not limited to use with the illustrated compression filter in which the output ~ n comprises the difference between the actual signal input fn and an estimated value ~n' Other compression filters may be used havingdifferenttransforms in which the compression filter output ~ is not a direct function of the difference between the actual input fn and an estimated value thereof, ~ . The use of the term "difference" signal values ~ n is intended to also identify the output from other suitable compression filters.
The compressed signal values ~ n are supplied, through switch 35, to an encoder 40 employing a truncated Huffman code for encoding the same. Huffman encoding is disclosed in U.S. Patent No. 4,396,906, issued August 2, 1983, entitled "Method and Apparatus for digital Huffman Encoding"
by Charles S. Weaver, which patent ls assigned to the same assignee as the present invention.
Brieflyt the Huffman encoding -technique makes use of the fact that the compression filter reduces the entropy of t:he signal output, ~ so that there can be a reduction Ln the total number of bits in the Huffman encoded signal over the input signal. A
single code word is assigned to infrequently occurring diffexence signals, and supplied as a label for the actual difference signal value A n.
In Fig. lA, the encoder 40 output is designated h( A ) and, at D in Fig. 2, the values h( A n)~

h( ~ n+1) etc. represent encoded values of ~ n-A n+1~ etc. The most frequently occurring value of A n (here zero) is encoded using the shortest code word. A truncated Huffman code is disclosed in U S. Patent Ilo. 4,396,906 which is readily implemented using a simple encoder and decoder. The encoder 40 output comprises code words for the most frequently occurring values of ~ n~ together with a combined code word label and actual value of the compressed signal ~n for less frequently occurring values of ~ n. For purposes of illustration, if the compressed signal value exceeds ~3 then the 0ctual compressed signal ~ n together with a code word label is produced at the encoder output. At Fig. 3, wherein several encoded compressed values are shown, it will be seen that the encoded value for ~ n+2 comprises a label together with the actual compressed signal A n+2~
wherein ~ n~2 comprises an infrequently occurring compressed signal value; that is, some value outside the rsnge of ~3.
The encoded signals from encoder 40 are recorded and/or transmitted to a remote receiver.
For recording, the encoder output is connected through a switch 48 to a recording unit 50 for recording of the encoded difference signals, labeled h( ~ n) signals. With the switch 48 in the other, broken line, position, the encoder output is supplied to a buffer memory 52 and thence to a digital modem 54 for transmission over transmission line 56. In certsin embodiments of this invention, check bits are generated for recording and/or transmitting along with encoded compressed signals -h( e, n). In some embodiments of the invention, digital input signals fn sometimes are supplied to 1~4~376 the input of the Huffman encoder through switch means 35, which signals serve to initialize, or reinitialize, the associated digital reconstruction filter described below.
Recorded encoded digital signals, such as those recorded at recording unit 50 of Fig. lA are reproduced using the system shown in Fig. ls, which system includes a playback unit 6Q. Recorded encoded digital signals from the playback unit 60 are supplied through switch 64 to a decoder 66 for decoding the truncated Huffman encoded signals. At the decoder 66, the Huffman code words are converted to the original compressed signals ~ n. Where the Huffman code word comprises a labeled actual compressed signal, the label is stripped therefrom, and the actual compressed signal without the label is supplied to the decoder output. Encoding and decoding means which may be used in the present invention are described in detail in the above-mentioned U.S. Patent No. 4,396,906. Coding and decoding are discussed in greater detail below under the heading "Encoding-Decoding".
The compressed signais ~ n from the decoder 66 are supplied to a linear reconstruction, or decompression, filter 70 through a buffer memory 72. The decoder output signals are produced at slightly varying rates, and the buffer memory 72 is included to accommodate the input rate requirements of the reconstruction filter 70. The reconstruction filter 70 converts the compressed signals A n to equal length sample signals fn(out) which closely match the input sample signals f to the compression filter 30. As noted above, one feature of this invention involves compression ~2'~4876 filtering without truncation and decompression filtering with truncation. In Fig. 2F, thl truncated reconstruction filter output fn(out), fn+1(out) etc. is shown to comprise words of 24 bits. Without truncation, the reconstruction filter would be required to handle word lengths of approximately 36 to 40 bits which, for consumer products, is not now feasible at reasonable cost.
~easons that suitable data compression with minimum distortion is obtained using a compression-decompression filter combination wherein compression filtering is effected without truncation and decompression filtering is effected with truncation will become apparent hereinbelow.
A digital to analog converter (D/A converter) 74 converts the signal samples fn(out) from the digital reconstruction filter 70 to analog form, for reproduction of the analog signals. A digital filter 75 which emphasizes the high frequency components of the signal output is included in the connection of the output from the digital reconstruction filter output to the D/A converter.
The frequency response of the filter 75 is shown in Fig 2A, adjacent the frequency response of the input filter 23. For simplicity, the same symbol fn(out) is employed at the input and output of the filter 75. Obviously, an analog filter having a similar frequency response may be included in the output from the D/A converter, in place of the digital filter 75. A receiver timing and control unit 76 supplies timing signals to the various receiver elements over line 78 for proper timing of the receiving operation. Also, control signals for the unit 76 are supplied thereto over line 80 from various elements of the receiver for control 4~37~;

thereof.
For transmission without recording, the encoded signals are transmitted over line 56 (from Fig. lA to Fig. lB) to a digital modem 82 at the receiver. The modem output is buffered by buffer memory 84, and the buffer memory output is supplied through switch 64 in the broken line position to the decoder 66 for decoding and subsequent processing in the manner described above.

QUANTIZATION OF ANALOG MUSIC SIGNALS
For purposes of illustration, and not by way of limitation, a music analog signal is considered for an input to the present system of this invention.
The entropy of a binary analog to digital (A/D) converted x bit long sample is H(q) = ~ - Pi log2Pi (2) ~ =l where there are 2X possible values of the sample and Pi is the probability that the ith possible value will occur. Let the size of the quantization level be q and, for simplicity, assume that the ith quantization, which gives the ith value, is from q(i-l) to qi. Then, as seen in Fig. 4, signal before analog to digital conversion will fall in the range q(i-l) to qi. In Fig. 4, the shaded area is the probability that the signal f(t) falls into the ith quantization.
Now, assume further that the size of the quantization level, q, is small compared to the standard deviation, ~ , of the analog music signal.

1~4876 If the A/D converter word length is increased by one bit, the quantization size is cut in half and, as shown by dashed lines in Fig. 4, two quantization bins are formed from the original bin.
For small q, the areas on either side of the vertical dashed line will be almost equal whereby the probability that f(t) will fall into one of the two new bins is approximately Pi/2. Therefore, the contribution of the two new bins to the entropy of the n+l bit long word vary nearly is r P P jl

-2~ ~ log2 f Pi = -21 2 (log2pi-l) = - Pi log2Pi + Pi (3) The entropy of the (x+l) bit long word is 2X 2X 5 H( 2 J =~ ~ Pilg2Pi + ~ Pi = H(q) + 1 (4) i=l ~=

From the above, it will be seen that as the bit length is increased, the increase in entropy will converge to one bit for each bit added to the word length. The argument when the first quantization bin is centered about zero (the usual case) is slightly more complex, however, the result is the same.
The Pi and entropies have been evaluated by numerical integration with various ratios of 6 to q for the Gaussian distribution. Table 1, be]ow, of calculated entropy increases at different ratios of ~ /q, shows that the entropies increase very closely to one bit each time q is cut in half or when the word length is increased by one.

Table l CAI.CIJLATED INCREASE IN ENTROPY AS
5QUANTIZATION SIZE IS RE`DIJCED

6 Entropy q (bits) 0.5 0.58 1 1.16 2 1.93 4 2.82 8 3,77 16 4.75 32 5,73 QUANTIZED MUSIC SIGNALS

The average word length of a Huffman encoder, such as encoder 40 to which the output from the compress:ion filter 30 is supplied is bounded as follows:

H(q) < average word length < H(q) + 1 (5) If a coefficient in the equation(s) that is used to realize the compression filter has a non-integer value the quanti~ation level at the filter output will be reduced; i.e., the minimum difference between possible output values will be decreased and H(q) w-.ll be increased. For example, 1~4876 consider the following two cornpressor equations to implement the compressioll filter transform.

~ n = fn ~ 2fn-1 ~ fn-2 (6) and n fn - 2fn_1 + 2 D~+lfn 1 +fn 2 - 2-m+1fn 2 +2-2mfn_2 (7) where m is a positive integer.
The Z-transform of Eq (6) has two zeros at (1,0) and Eq (7) has two zeros at (1-2-m, 0) in the Z-plane. The ~ n in Eq (7) will have values spaced a distance of 2 2mq. When m is large, the 6 from both filters will be approximately equal but the ~ /quantization level ratios will differ by a factor of 2-2m. Therefore, the entropy of Eq (7) will be approximately 2m more bits than the entropy of Eq(6), and after Huffman encoding the average bit length will be approximately 2m bits longer.
It will be noted that multiplying the right side of Eq(7) by 22m returns the quantization level to q, but the standard deviation is increased by a factor of 22m so that the ratio is unchanged.

Filter Weights Versus Word Length A general form of a compression filter difference equation is Q

n = ~ aifn-j+l i =l where ai is a constant. If ai can be represented by a finite length binary number, it can be expressed as 12~87fi ai = ~-~ bij2i , (9) where bij = 0 or +1 and j can have positive or negative values. Any negative j and a non-zero bij means that fn 1 is shifted right and added; jO
bits must be added to the least-significant end of the arithmetic word, where jO is the most negative j with non-zero bij.
The Z-transform of Eq (6) is G(Z) = (1 _ 2z-l + z-2) = (1 _ z-1)2 (10) which can be represented by two zeros at (1,0) in the Z-plane as seen in Fig. 5. The frequency response at fO of an all-zero digital filter [e.g., Eq(6) or Eq(7)] is the product of the distances from the point exp (j2~T fT) to each of the zeros and the gain constant [equal to 1 in Eq(10)], where f is the frequency and T is the time between samples. Thus, the frequency response of Eq(10) i s : .
R(fo) = d2 (11) If there are n zeros at (1,0), R(fo) = dn (12) These compression filters reduce the entropy for the following reason: if the A/D sampling rate is 44 x 103 samples per sec, the point (-1,0) on the unit circle corresponds to a frequency of 22 KHz.
The centroids of the music spectra usually will be less than 1 KHz, so that most of the spectral points correspond to points on the unit circle that lX~487~i are near (1,0). The vslue of d (and dn) will be much less than one, and the integral of the spectrum value time dn as B function of e (where ~ , 2q~ f) will be less thsn the variance of the input spectrum (R~l). A reduced var~ance means a reduced entropy.
The value of d is greater than one when ~ >60 (f ~ 7.33RHz) so that spectral components sbove 7.33 KHz are amplified by dn for filters with all of the zeros at (1,0). There is a value of n such that increasing n above this value amplifies the total energy above 7.33 R~z more than the total energy below 7.33 RHz is attenuated. This value of n minimizes the output variance and the entropy because the input and output q are the same when K = 1. This can be seen as follows:
G(~) =K(l _ z-l)n n =K ~ n! (-1) _ z-i (13) ~__ i! (n-i)!
i=no eK~ aiz i i =O
where the ai are the constants that are used in Eq(~), and K is the gain constant. The ai are integers that can be expanded [as in Eq (9)]
without negative j and, therefore, q is not reduced.
Note that different values of R do not change the ratio ~ /q or the entropy when K is a power of two, because the input word is only shifted. Thus, the n that minimizes the entropy for R ~ 1 also minimizes it for other R and the minimum entropy is obtained when K is a power of two.

Two other zero positions on the unit circle that do not reduce q sre at (-1,0) and (the complex pair) at (0,1) and (0,-1). The Z-transforms are:

G(Z) ~ (1 + z~l)n [n zeros at (-1,0)] and G(Z) c (1 + z-2)n (15) [n zeros at (0,1) and n zeros at 0, -1].
The two other complex-pair positions that do not change q have the following transforms:

G(Z) = (1 _ z-l + z-2 )n (16) (which places n zeros on the unit circle at angles of +60 from the origin and n at -60) snd G(Z) ~ (1 + z-l + z-2 )n (17) 15 (which p:Laces n zeros at + 120 and n at -120).
The above are the only zero positions inside or on the unit circle that do not decrease q. There are none outside the uni~ circle that result in a satisfactory reconstruction filter.
One other zero position that is of interest for music data compression is the complex pair at 41.41 on the unit circle. The transform is G(Z) ~ .5z~l + z-2. (18) This angle corresponds to 5.06 ~Hz and q is divided by 2 for each complex pair.

12'~A876 In accordance with one aspect of the present invention a compression filter is employed in which the Z-transform thereof has zeros on the unit circle at at least one of the above-identified complex-pair positions (i.e. ~41.41, ~60~, ~90, +120 and 180).
In addition to these zero positions, the compression filter may also have one or more zeros on the unit circle of the Z-transform at zero degrees. The above-described zero positions are shown in Fig. 6.
As noted above, these zero positions on the unit circle minimize the entropy, and the combination of zero positions employed is dependent upon the spectrum of the signal to be compressed. For example, zeros can be placed at the ~60 points to reduce the part of the output variance that is due to high frequencies (from say 3 to 14KHz~ so that more zeros can be used at 1,0. In Fig. 7, the frequency response of three different compression filters is shown which filters have zeros on the unit circle of the Z-transform at OQ; OQ and ~60; and 0, ~90 and ~120. It will be seen that the useable zero positions for entropy minimization allow for design of compression filters having a wide range of frequency responses.
Of course, limitations are encountered when the number of ze!ros is increased. The amount of calculation required is directly proportional to the number of zeros, and the filter arithmetic word length increases by at least one for each additional zero. Also, recovery during reconstruction from bit errors will take longer as the number of zeros is increased.
After the compression filter transfer function has been chosen, the entropy that will be obtained can be estimated as follows: the music spectrum S(f~ is measured, and the integral 12~87~

¦G(Z) ¦2 s ( e;i2~fT) d~ (19) is integrated along the unit circle fron' (1,0) to (-1,0), where S~ ( ~j2~ fT)= S(f) .

The square root of the integral is the ~ of the compression filter output. Table 1 now can be used to estimate H(q).

Compression Filter Although it will be apparent that standard digital techniques may be used for implementing the above described compression filter transforms, including the use of a programmed digital computer, a block diagram of a second order digital compression filter suitable for use in implementing equation (6) is shown at Fig. 8~ to which figure reference now is made. The illustrated compression filter includes a series of shift registers 102, 104, and 106 into ~hich consecutive sample si&nals from the A/D converter, through filter 23, are shifted. In Fig. 8, for purposes of description, the registers, 102, 10~ and 106 are shown to contain samples fn~ fn 1~ fn 2~ respectively. For 14-bit samples, 14-bit registers are employed. The register outputs are connected to a digital multipl~exer 108 for selective connection of the sample signals to an arithmetic and logic unit (ALU) 110. The multiplexer 108 and ALU 110 are under control of timing and control unit 24.
As noted above in the description of Fig. lA, the digital compression filter 30 may include an estimator 32 having an output comprising an 4~7fi 2~

estilllated sample vslue ~n hased upon actual sam~Jles fn 1 and fn+1 occurring hefore and after the samp]e fn to be estimated. Often, prior art estimators are used which provide an output, fn = alfn+l + a2fn-1 (20) where tht-~ coefficients al and a2 are chosen to minimize the mean square error of the difference n~ here ~ n = fn ~ ~n~ as noted in equation (1), above. Ft)r al = a2 = 1, equations (1) and (20) may bt-~ combined to give ~ n = fn+l - 2fn + fn-l (21) (It here will be noted that equations (6) and (21) are equivalent.) Fquation (21) may be utilized by the illustrated compression filter in the generation of the compressed signal ~n An estimate fn of the sample fn is made using the samples either side of n n-l and fn~ Dut not fn itself. Under control of unit 249 the words fn 1 and fn+l are moved :into the ALU 110 through the multiplexer 108 and adtled. The actual sample fn then is moved into the AL.U 110 through the multiplexer 108, and multiplied by 2. ~iultiplying by 2 simply involves shifting of the bits toward the most significant bit. The actual sample fn~ multiplied by 2, is subtracted from ~n to provide the compressed signal value f~n at the ALU 110 output, which then is supplied to encoder 40. The arithmetic in the ALU
110 is done in a word length sufficiently long to ensure against truncation or round-off error. It will be seen that data compression by the above-lZ~4~

described appsratus includes estimating a ssmple~alue by interpolation.

HUFFMAN ENCODING AND DECODING
As noted abo~e, Huffman encoding and de~od$ng means suitsble for use in the present Rystem for encoding and then decoding the compression filter output are disclosed ln U,S. Patent Number 4,396,906 entitled, "Method and Apparatus for Digital Huff~an Encoding" by the precent inventor-Reference is ~ade to Fig. 9 wherein ane~smple of n truncsted Huff~an cote is rhown for purposes of illustratio~ only and not by way of limitatlon. Tbere, a table of compressed signals, ~n~ rsnging from ~3 is shown together with u code word for said ~ignal6,the length of the code word, snd the relative probabilley of ocrurrence of aid compressed ~ignals. The compressed Ai~nsls ~ n which occur most frequently (here those between +3) are assigned a code ~ord. The probhbility of ~ n comprising a ~alue which ls assigned a code word is high, ~ay, .9~. These compressed signals are assigned different lenglth code words, with the most frequenltly occ~rring compressed signal being 8ssigned the ~hortest code word. In the table, the most frequently occurring compressed Bignal~ ~ n'~
is as6igned the shortest code word, and the least frequently occurring compressed signal, ~ n --3, is assigned the longeslt code word. All other compressed signals outside the range of ~3 are-identified as else in ithe table, and these are assigned a code word which, as tescribed above ~ith 1~4~7fi reference to Figs. 2 and 3, comprises a label for the flctual compressed signal value a n which subsequently is transferred to the ~luffman decoder by way of recording, transmission over a communications link, or the like. Obviously, the system is not limited to use with the illustrated truncated Huffman code. Additional compressed signals, ~ n' may be assigned a code word, and other code words may be employed.

RECONSTRUCTION FILTERING

Entropv Versus Reconstruction Filter Stability .
For exact reconstruction of the digital music signal supplied to the digital compression filter 30, the reconstruction filter 70 transfer function ]5 would have to be the inverse of the compression filter 30 transfer function. (Two other necessary conditions for exact reconstruction are that there be no over- or under-flow errors in the filter arithmetic and that there be no truncation of the compression filter output word length ) As shown above, minimum entropy is obtained when the zeros of the compression filter transfer function are on the unit circle. The exact inverse has poles on the unit circle in the same positions as the compression filter zeros. Such a reconstruction filter is unstable. Such instab:ility is satisfactory until a bit error occurs whereupon incorrect, and random, "initial conditions" cause the reconstruction filter to diverge to saturation. Two different systems are described hereinbelow for use with arrangements wherein the compression-reconstruction filter combination is unstable which systems provide for lX~4876 recovery from bit errors.
Briefly, one system includes the periodic transfer of a plurality of actual digital slgnal values, f , to the digital reconstruction filter 70 to periodically reinitialize the same. This, of course, requires blocking of the signal and, without very complex and highspeed logic, the loss of data from the point of the error until the end of the block.
Another system includes the use of check bits and error checking means for production of a bit error signal whenever an error is detected. The error signal is used to momentarily move the poles of the reconstruction filter 70 inwardly of the unit circle, during which time the filter 70 recovers from errors without the need to reinitialize the filter with actual signal values f . By locating the poles of the reconstruction filter inside the unit circle, the filter is stable and incorrect "initial conditions", due to errors, will damp out.
Under these conditions, the filter is stable and no blocking is required for recovery from errors.
Stable filter combinations of this type also are described in further detail hereinbelow.
Compression Filter Zeros Inside Unit Circle Unsatisfactory It here will be noted that poles of an exact inverse are off the unit circle if the compression zeros are not on the unit circle. However, such a compression-reconstruction filter combination is not practical for music data compression as will become apparent from the following example. If the compression filter transfer function contains two real zeros near the 1,0 point, they must be aL a distance from the unit circle that is less than 37~

`~ x 200/22000 (the distance around the unit circll that corresponds to 200 ~Iz). A larger distance means that the low frequency components will not be as highly attenuated. Since 0.00909~ ~- 2-7, the zero ~ould be [ 1 ( 1 2-7) -1]

Therefore, if two zeros were used, there would be a coefficient equal to 2-14, and 14 bits would be added to the least significant end of the filter arithmetic. With no truncation, approximately 14 bits would be added to the entropy and little data compression would be possible.
Another compression-reconstruction filter combination which is not suitable for use in the present system, also includes arrangements wherein both the zeros of the compression filter and the poles of the reconstruction filter are off the unit circle. With these arrangements there is no arithmetic word length truncation, however, the output of the compression filter is truncated to a length that is one or two bits longer than the analog l:o digital converter word length. Letting the out:put quantization level equal qO1 the quantizing noise power will have a variance of ~ 2 = qo (22) Io 12 and the noise will be white. The reconstruction distortion will be equal to the output noise due to a noise generator at the reconstruction filter input that has a variance that is given by equation (22) Since the input noise samples are white and statistically independent, it can be shown that the output noise varlance is qO ~ gi (23) or ~ = ql ~ gi ) (24) where gi is the ith value (at the ith sampling time) of the impulse response of the reconstruction filter. In other words, the square root of the sum-of-the-squares of the impulse response samples is the standard deviation multiplier. This multiplier has been calculated for different pole positions by solving the appropriate difference equation. From the calculations it has been determined that noise power produced by such truncation of the compression filter output is too large, or is concentrated within such a small portion of the signal bandwidth, so as to produce undesirable sound in the music output.
Consequently, truncation of the compression filter output words is unsatisfactory for music data compression.

Reconstruction Filter Not an Exact Inverse of Compression Filter If there are no bit errors in the transfer of the compression filter 30 output to the input of the digital reconstruction filter, and no truncation of the compression filter 30 output, the output from the ~luffman decoder 66 is identical to 487~i the output from the compression filter 30. Thus, it will be understood that the transfer from the input to the compression filter 30 to the output of the reconstruction filter 70 is simply the product of the transforms of the two filters 30 and 70.
Another data compression system which embodies the present invention includes a compression-reconstruction filter combination wherein the zeros of the compression filter are at specific points on the unit circle to reduce the entropy, and corresponding poles of the reconstruction filter are located inside the unit circle, adjacent said zeros for stability. The frequency response and stability of such compression-reconstruction filter combinations are readily calculated. Consider, for example, a compression-reconstruction filter combination wherein the compression filter has two zeros at (1,0) and the reconstruction filter has two poles at (1-.00195, 0). The pole-zero pattern of such a compression filter cascaded with a reconstruction filter is shown in Fig. 10, and the frequency response of the filter combination is shown in Fig. 11. As seen in Fig. 11, the combination provides a very flat high-pass filter with a 18Hz cut-off. With this filter combination, recovery from bit errors is within 20-30ms. It here wil] be noted that the reconstruction filter 70 employed herein preferably comprises a digital computer programmed for the desired reconstruction filter operation.
_ord Length Considerations in the Reconstruction_ _ _ _ _ _ _ _ _ _ _ __ _ _ _ _ _ _ _ _ _ Filter A stable reconstruction filter, operating without truncation, would require a large arithmetic word length. For example, the above-12;~487~

describc(l-lcal-polc 18 1l~ filter woul(l rCq~lirC ~t I e.lst 34 bit arithmetic (.0()l95 = 2 9), at: 1cast 9 hits per pole on the 1ecl.st ~significant end an(l I
bit on the most sign:ificant end per pole when the analog to digital (A/D) word length is 14 bits. 4-pole configurations would require even longer arithmetic word length. Presently, computcrs operating with such large word lengths are not practical for consumer music data compression.
Fortunately, the reconstruction filter arithmetic words can be truncated to practical lengths with negligible system degradation.
Arithmetic word truncation noise is analyzed in substantially the same manner as is the analysis of quantizing. For this analysis, noise generators with noise power q2/12 (one generator for each coefficient) are added to the filter input and the input-to-output standard deviation multiplier is calculated. The value of q is the quantization level of the truncated arithmetic word.
For an 18 Hz 2-real-pole reconstruction filter, the multiplier is 3227. Then, the arithmetic word length must be 12 bits longer than the A/D word length or the arithmetic truncation noise power will be larger than the A/D quantizing noise power. Fourteen bit A/D conversion requires 26 or 27 bit reconstruction filter arithmetic. If the compression filter has two real zeros at (1,0) and two complex poles at 7KHz on the unit circle, then a reconstruction filter having a complex pair of poles at the 20Hz Butterworth position and a complex pa-ir 7.33 ~Hz along the unit circle and lOOHz in from the unit circle may be used. The standard deviation multiplier of such a reconstruction filter is 32, or 5 bits, and the 1~48~

frequency response of the compression-reconstruction filter combination is substantially the same as that shown in Fig. 11, except for a narrow notch at 7~H~. With 24-bit arithmetic there is virtually no degradation of the signal. Digital computers with, say, 24 bit arithmetic for reconstruction filtering are available at a reasonable cost for use in the system of this invention.

SYSTEM WITH ERROR DElECTION AND
MOVABLE RECONSTRUCTION FILTER POLES
In Figs. 12A and 12B, a modified form of this invention is shown wherein check bits are generated for recording and/or transmitting along with the encoded digital compressed signals. At the playback and/or receiver unit, any errors detected using the check bits serve to generate an error signal which is used to momentarily move the poles of the digital reconstruction filter inwardly, or further inwardly, of the unit circle in the z-plane without changing the pole angle. For an ~nstable reconstruction filter, momentary movement of the poles inwardly of the unit circle results in a stable filter combination which recovers from playback and/or transmission errors without the need for reinitialization of the filter. For a stable reconstruction filter, momentary movement of the poles inwardly of the unit circle provides for accelerated recovery from error signals.
Reference first is made to Fig. 12A wherein the digital recording and transmitting portion of a modified form of data compression system which includes the use of check bits is shown. The system of Fig. 12A is similar to that of Fig. lA

and is shown to inc1ude an analog to digital converter 20, digital compression filter 30, Huffman encoder 40, switch 48, recorder 50, buffer memory 52, modem 54 and timing and control unit 24, all of which may be of the same type as shown in Fig. lA and described above, lt will be noted that an analog high frequency deempha sis filter 23A is included in the input of the A/D converter which filter serves the same function as digital filter 23 shown in Fig. lA.
In the form of invention shown in Fig. 12A, a check bit generator 90 is shown included in the connection of the Huffman encoded signal h( ~ n) to the recorder 50 or modem 54, dependent upon the position of switch 48. Check bits generated by check bit generator 90 are added to the digital stream of Huffman encoded signals for recording and/or transmission along with said encoded digital compressed signals. Numerous schemes for the generation of check bits, and for error detection using such check bits are well known and require no detailed description. It here will be noted that recorders and modems often include check bit generator means for the generation of check bits to be added to the data stream to be recorded, or transmitted.
Recorded encoded digital signals, with check bits, such as those recorded at recorder 50, are reproduced using playback unit 60 shown in Fig.
12B, to which figure reference now is made.
Signals transmitted by modem 54 ~Fig. 12A) are transmitted over line 56 to modem 82 (Fig. 12B).
Switch 64 connects the playback output, or modem outut, to an error checking circuit 92 where the signal stream is checked for bit errors. When an 4t~

error is detected, a bit error signal is generated which slgl-a1 is transmitted over line 94 and through switch 96 to the digital reconstruction filter for momentarily shifting the poles of the filter inwardly.
Check bit signals are stripped from the signals from the playback unit 60 and/or modem 82 by the error checking means 92, and the Huffman encoded digital compressed signal stream h( ~ n) from the error checker is supplied to the Huffman decoder 66, which decoder may be of the same type as shown in Fig. lB and described above. From the Huffman decoder, the digital compressed signals ~ n are supplied through buffer memory 72 to the digital reconstruction filter 70A. As with reconstruction filter 70 of Fig. lB, the reconstruction filter 70A operates with truncation and converts the compressed signal input~n thereto to equal length sample signals fn(out) which closely match the input sample signals fn to the compression filter 30 (Fig. 12A). A digital to analog converter 74 converts the signal samples fn(out) to analog form, f(t) out. An analog high frequency emphasis filter 75A is included at the output of the D/A converter, which filter serves the same function as filter 75 in Fig. lB; i.e. to restore the amplitude of the high frequency signals which were deemphasized by filter 23A.
Poles Inside Unit Circle in Z-plane As noted above, one embodiment of the present invention includes the use of a compression-reconstruction filter combination wherein the zeros of the compression filter are at specific points on the unit circle to reduce entropy, and the reconstruction filter has corresponding poles i2X4876 inside the unit circle, adjacent said zeros which provide for stable operation. Recovery of the reconstr llC tion filter from bit errors is accelerated by momentarily moving the poles of the reconstruction filter inwardly of the unit circle in the Z-plane whenever an error signal is produced by error checker 92.
Reference is made to Fig. 13 wherein zeros and poles of the transfer function of a compression-reconstruction filter combination are shown. The compression filter zeros are shown on the unit circle at zero degrees, and a pair of reconstruction filter poles are shown adjacent the zeros and normally at a distance of 0.00195 inside the unit circle. It will be noted that this combination of zeros and poles is the same as that shown in Fig. lO, described above. Ilowever, in the system illustrated in Fig. 13, the reconstruction filter poles are momentarily moved inwardly upon receipt of a bit error signal from the error checker 92 over line 94. For purposes of illustration, the poles are shown moved to a point 0.0625 inside the unit circle for rapid recovery from the error. After a short period of time, say 50 ms, the poles of the reconstruction filter return to normal position, that is, to a point 0.00195 inside the unit circle at zero degrees.
Difference equations for a reconstruction filter having two poles at 0 and inside the unit circle are:

Yn 2~ n + aYn-l = 2~ n + Yn_1 ~2-myn_l (25) fn = Yn + afn-l = Yn ~~ fn-l -2 fn-l (26) ~2;~4~7~, where: a = 1-2-m and m = an integer.
~ reconstruction filter for implementing equations (25) and (26) is shown in Fig. 14, to which figure reference now is made. The illustrated reconstruction filter 70A, comprises a 4 to 1 digital multiplexer 130 having one input 132 to which compressed signals ~ n are supplied from the decoder 66. The output from the multiplexer 130 is supplied to an arithmetic and logic unit, ALU, 134 where the required multiplication by shifting, addition and subtraction take place under control of timing and control unit 76A.
The output from ALU 134 is connected to the input of a 1 to 2 digital demultiplexer 138. One output of the demultiplexer 138 is connected to one register of a pair of series connected shift registers 140 and 142 over line 144. The other demultiplexer output is connected over line 146 to a single shift register 148. The value of Yn determined by the ALU is loaded into register 140 while the prior value of Yn is shifted from register 140 into register 142. The third register 148 is supplied with the sample value fn(out) as calculated by the ALU 134.
Outputs from registers 140, 142, and 148 are supplied as inputs to the ALU 134 through the multiplexer 130. When used, the value stored in register 148 comprises fn l(out). From equation (25) it will be seen that the value Yn is calculated using the ~ n and Yn 1 inputs to ALU 134 available at line 132 and from register 142. From equation (26) it will be seen that the sample value fn(out) is calculated using the Yn and fn l(out) inputs from registers 140 and 148, respectively.

So long as n~ equals an integer less than infinity, the reconstruction filter 70A operates s t a b 1 y , a n d n e i t h e r i n t i a l i z a t i o n n o r reinitialization of the filter is required. In the 5 absence of bit errors the filter is operated with a relatively large value of m, say m=9, to place the poles of the filter adjacent the unit circle at .00195 from the unit circle. When an error is detected by error checker 92, a smaller value of m 10 is used, say m = 4, thereby moving the poles of the filter inwardly to a point .0625 from the unit circle. The bit error signal from error checker 92 (Fig. 12B), which is supplied to the ALU 134 over line 94, controls the value of m used in the 15 implementation of equations (25) and (26) simply by controlling the amount of shifting to perform the indicated multiplications by the factor 2-m. When an error is detected, contents of an ALU register are not shifted as far to the right for som e 20 nominal length of time (say 50 ms) when performing the multiplications by 2-m, thereby moving the reconstruction filter poles inwardly away from the unit circle to accelerate recovery from transients.
After this short time period, operation returns to 25 normal with the reconstruction filter poles again adjacent the unit circle.

Poles Normally _ Unit Circle in Z-plane As noted above, another embodiment of the present invention involves use of a compression-30 reconstruction filter combination wherein the zerosof the compression filter are at specific points on the unit circle to reduce entropy and the reconstruction filter has corresponding poles which also are on the unit circle at the same locations 1~24876 as the zeros, during normal operation; that is during operation in the absence of bit errors.
~lowever, when a bit error is sensed by error checker 92, the poles of the reconstruction filter 70A are momentarily moved inwardly of the unit circle for stable reconstruction filter operation, and recovery from the error. This embodiment may be implemented using the above-described receiving, or playback, unit shown in Fig. 12B, and reconstruction filter 70A shown in Fig. 14. Now, however, the reconstruction filter 70A, in the absence of transients, operates with poles on the unit circle, as shown in Fig. 15. In Fig. 15, two zeros of the compression filter 30 are shown located on the unit circle in the Z-plane at zero degrees, and, during operation in the absence of bit errors, the two poles of the reconstruction filter 70A are located at the same point on the unit circle, at 2 point where m equals infinity.
In the presence of a bit error signal from error checker 92, the reconstruction filter poles are momentarily moved inwardly, of the unit circle, at zero degrees. For purposes of illustration, the value of m is shown changed to 4. Under these conditions, the reconstruction filter quickly recovers from the error without the need for initialization, or reinitialization of the filter by the transmission of actual signal values fn thereto. It here will be noted, that at the start of operation, the reconstruction filter poles are momentarily moved inwardly of the unit circle to avoid generation of a random ramp function at the output therefrom.
Obviously, the invention is not limited to inward movement of reconstruction filter poles to a ~ ~4~76 sing]e location in the presence of an error signal.
Several values of m msy be employed, with filter operation being stepped through several different pole locations during recovery from bit errors.
For example, values of m equal 2, 4, and 7 may be used wherein operation first is switched to m equals 2, then m equals 4, and finally to m equals 7, before stepping back to the original value of m, either on, or inside, the unit circle in the Z-plane.

SYSTEM WITH PERIODIC TRANSFER OF fn Another embodiment of the present inventioninvolves use of a compression-reconstruction filter combination wherein the zeros of the compression filter are located at specific points on the unit circle in the Z-plane to reduce entropy, and the reconstruction filter has corresponding poles on the unit circle at the same locations as the zeros, which poles are fi~ed and are not moved inwardly.
As noted above, such a compression-reconstruction filter combination is unstable, and, any transients result in a random output from the reconstruction filter. To minimize the effects of any such transients, the reconstruction filter is periodically reinitailized during operation by transfer thereto of a plurality of actual signal values, fn. Apparatus illustrated in Figs. lA and lB may be used for this type of operation.
The signal stream which is transferred by this operation is illustrated in Fig. 16, to which figure reference now is made. In addition to Huffman encoded difference signals h( ~ n+2) h( ~ n+i)~ Huffman encoded signal values h(fn)~
h(fn+l), etc. are periodically transmitted, by 12~4876 periodic actuation of switch 35 to the broken ]ine pOSitiOIl shown in ~ig. 1~ ith switch 35 in the broken line posit-ion, a series of actual signaL
values, f are pericdicnlly supplied to the liuffman encoder 40 for encoding and subsequent recording and or transmission. The number of consecutive signal values, f , sent equals the order of the reconstruction filter.
In the signal stream shown in Fig. 16, two consecutive signal values f are periodically encoded, for use in periodically reinitializing a second order reconstruction filter 70. The encoded signals h(f ) etc. are shown to comprise a label and the actual signal value f . The label used is different from the "else" label used to identify those signals outside a predetermined range of compressed signal values ~ . The label portion of the encoded signals h(f ) are designated Label # 2 in Fig. 16 to distinguish from the "else" label.
The required number of encoded signal values h(f ) are periodically transferred, say, every 10 milliseconds, as shown in Fig. 16 to periodically reinitialize the associated digital reconstruct:ion filter 70. With this periodic reinitialization of the reconstruction filter, there is no requirement to operate the reconstruction filter with poles inside the unit circle in the Z-plane since any ramp function output produced by transient signals is eliminated within 0 to 10 milliseconds time.
The invention having been described in detail in accordance with requirements of the Patent Statutes, various other changes and modifications will suggest themselves to those skilled in this art. For example, many of the illustrated functions may be implemented using a digital ~Z~4~37~.

computer with suitable computer routines. It is intended that this and other such changes and modifications shall fall within the spirit and scope of the invention defined in the appended S claims.

Claims

1. In a data compression system for processing a stream of fixed length digital sample signals, the combination including, linear digital compression filter means responsive to the stream of digital sample signals for generating a stream of compressed signals, the transfer function of the digital compression filter means having zeros on the unit circle in the Z-plane at at least one of the following angles measured from the origin; +41.41°, +60°, +90°, +120° and 180°, digital encoding means responsive to the stream of compressed signals from the digital compression filter means and implementing a variable word length code, digital decoding means, means for transferring the output from said digital encoding means to said digital decoding means for decoding the same, and linear digital reconstruction filter means responsive to the output from said digital decoding means for reconstruction filtering of the output therefrom, the transfer function of said digital reconstruction filter means having poles on or inside the unit circle in the Z-plane at substantially the same angular positions as the zeros of the digital compression filter means.

2. In a data compression system as defined in Claim 1 wherein the digital compression filter means performs arithmetic operations without truncation or round-off, and the digital reconstruction filter means performs arithmetic operations with truncation for production of truncation errors in the output therefrom.

3. In a data compression system as defined in Claim 1 wherein said transferring means includes use of an error checking code and error detecting means for detection of errors in the transfer of the output from the digital encoding means to the digital decoding means, and for generation of an error signal when an error in the transfer is detected, means responsive to an error signal from said error detecting means for momentarily moving poles of said digital reconstruction filter means inwardly of the unit circle in the Z-plane without changing the pole angle to facilitate recovery of the digital reconstruction filter means from detected errors.

4. In a data compression system as defined in Claim 3 wherein the transfer function of the digital reconstruction filter means has poles on the unit circle in the Z-plane, which poles momentarily are moved inside the unit circle in response to an error signal from said error detecting means.

5. In a data compression system as defined in Claim 3 wherein the transfer function of the digital reconstruction filter means has poles inside the unit circle in the Z-plane, which poles momentarily are moved further inside the unit circle in response to an error signal from said error detecting means.

6. In a data compression system as defined in Claim 1 wherein the transfer function of the digital reconstruction filter means has poles on the unit circle in the Z-plane at angular positions substantially corresponding to the zeros of the transfer function of the digital compression filter means, and means for periodically supplying a plurality of successive digital sample signals to the digital reconstruction filter means for periodically reinitializing operation thereof.

7. In a data compression system as defined in Claim 6 wherein said digital sample signals which are supplied to the digital reconstruction filter means are supplied thereto through said digital encoding means, digital decoding means, and said transferring means.

8. In a data compression system as defined in Claim 7 wherein said digital encoding means is periodically operated for periodically labelling a plurality of successive digital sample signals for transfer to said digital reconstruction filter through said transferring means and digital decoding means.

9. In a data compression system as defined in Claim 8 wherein said digital encoding means is periodically operated every 6 to 16 milliseconds for periodically supplying the digital reconstruction filter means with successive digital sample signals.

10. In a data compression system as defined in Claim 1 wherein the compressed signals from the digital compression filter means are related to the difference between the sample signal input thereto and an estimated value thereof, said estimated digital sample signal value being obtained using sample signals from both sides of the digital sample signal to be estimated.

11. In a data compression system as defined in Claim 1 including analog to digital converting means from which digital sample signals are obtained by analog to digital conversion of analog signals, and digital to analog converting means for converting digital output signals from said digital reconstruction filter means to analog form.

12. In a data compression system as defined in Claim 11 wherein said analog signals comprise music signals.

13. In a data compression system as defined in Claim 12 wherein digital sample signals are obtained from said analog to digital converting means at a rate of between 30 to 50 KHz.

14. In a data compression system as defined in Claim 1 wherein said transferring means comprises, means for recording the encoded signal from said digital encoding means, and means for playback of the signal recorded by said recording means.

In a data compression system as defined in Claim 1 wherein said transferring means comprises first and second modems and a transmission link interconnecting said modems.

16 In a data compression system as defined in Claim 1 including an input filter for high frequency deemphasis of the signal input to said digital compression filter means, and an output filter for high frequency emphasis of the signal output from said digital reconstruction filter means.

17, In a data compression system as defined in Claim 16 wherein said input and output filters are of the digital type.

18. In a system for digital data compression which includes a source of fixed length digital data sample signals, linear compression filter means for generating compressed signals, linear reconstruction filter means responsive to compressed signals from said compression filter means for reproducing said digital data signals, and means for transferring the output from the compression filter means to the input of the reconstruction filter means, a method of operating said compression and reconstruction filter means to provide a reduction in signal entropy with little signal distortion, the improvement including operating the digital compression filter means with a transfer function having zeros on the unit circle in the Z-plane at at least one of the following angular positions measured from the origin, +41.41°, +60°, +90°, +120°, and 180°, and operating the digital reconstruction filter means with a transfer function having poles on or inside the unit circle in the Z-plane at the same angular positions measured from the origin as the zeros of the digital compression filter means.

19, In a method as defined in Claim 18 including, operating the digital compression filter means without signal truncation, and operating the digital reconstruction filter means with arithmetic word length truncation for production of truncation errors in the output from the digital reconstruction filter means.

20. In a method as defined in Claim 18 including, momentarily moving the poles of the reconstruction filter means inwardly away from the unit circle in the Z-plane in response to transient errors in the transfer of the output from the compression filter means to the input of the reconstruction filter means to accelerate recovery of the reconstruction filter operation from said errors.

21. In a method as defined in Claim 18 including, operating the digital reconstruction filter means with poles on the unit circle in the Z-plane, and periodically supplying a plurality of successive digital sample signals to the reconstruction filter means for periodically reinitializing operation thereof, the number of successive digital sample signals supplied to the reconstruction filter means being equal to the order of said reconstruction filter means.

22. In a data compression system for handling a stream of digital sample signals, the combination comprising, linear digital compression filter means responsive to the digital sample signals for generating compressed signals, the transfer function of the digital compression filter means having zeros on the unit circle in the Z-plane at at least one of the following angles measured from the origin; +41.41°, +60°, +90°, +120°, and 180°, linear digital reconstruction filter means having a transfer function with poles on or inside the unit circle in the Z-plane at substantially the same angular positions as the zeros of the digital compression filter means, and means for transferring the output from the digital compression filter means to the digital reconstruction filter means.

23. In a data compression system as defined in Claim 22 wherein said transferring means includes Huffman encoder means for truncated Huffman encoding of the output from the digital compression filter means, and means responsive to the output from the encoder means for decoding the same.

24. In a data compression system as defined in Claim 22 wherein the transferring means includes bit check generator means and means for generating an error signal when a transient error is present, and means under control of said error signal for momentarily moving the poles of the reconstruction filter means inwardly of the unit circle without changing the angular position thereof for accelerated recovery from detected transient errors.

25. In a data compression system for handling a stream of digital sample signals, the combination comprising, digital compression filter means responsive to the digital sample signals for generating compressed signals, the transfer function of the digital compression filter having zeros on the unit circle in the Z-plane, digital reconstruction filter means having a transfer function with poles on or inside the unit circle in the Z-plane at substantially the same angular positions as the zeros of the digital compression filter means, means for transferring the output from the digital compression filter means to the digital reconstruction filter means, and means responsive to transient errors in the transfer of the output from the digital compression filter means to the digital reconstruction filter means for momentarily moving poles of the reconstruction filter means inwardly of the unit circle in the Z-plane without substantially changing the angular position of said poles to facilitate recovery from transient errors.

26. In a data compression system for processing a stream of fixed length digital sample signals, the combination including, linear digital compression filter means responsive to the stream of digital sample signals for generating a stream of compressed signals, the transfer function of the digital compression filter means having zeros on the unit circle in the Z-plane, linear digital reconstruction filter means, means for transferring the output from the digital compression means to the digital reconstruction filter means, said digital reconstruction filter means having poles on the unit circle in the Z-plane at angular positions corresponding to the zeros of the transfer function of the digital compression filter means, and means for recurrently supplying a plurality of successive digital sample signals to the digital reconstruction filter means for recurrently reinitializing operation thereof.

27. In a data compression system as defined in Claim 26 wherein said means for recurrently supplying a plurality of successive digital sample signals to the digital reconstruction filter means includes said transferring means.

28. In a data compression system as defined in Claim 26 wherein said means for recurrently supplying a plurality of successive digital sample signals to the digital reconstruction filter means is operated periodically for periodically reinitializing operation thereof.