Classifications

• • H—ELECTRICITY
  • H03—ELECTRONIC CIRCUITRY
  • H03G—CONTROL OF AMPLIFICATION
  • H03G7/00—Volume compression or expansion in amplifiers
  • H03G7/007—Volume compression or expansion in amplifiers of digital or coded signals
• • H—ELECTRICITY
  • H03—ELECTRONIC CIRCUITRY
  • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
  • H03H17/00—Networks using digital techniques
  • H03H17/02—Frequency selective networks
  • H03H17/0223—Computation saving measures; Accelerating measures
  • H03H17/0227—Measures concerning the coefficients
  • H03H17/023—Measures concerning the coefficients reducing the wordlength, the possible values of coefficients
• • H—ELECTRICITY
  • H03—ELECTRONIC CIRCUITRY
  • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
  • H03H17/00—Networks using digital techniques
  • H03H17/02—Frequency selective networks
  • H03H17/0248—Filters characterised by a particular frequency response or filtering method
  • H03H17/0264—Filter sets with mutual related characteristics
• • H—ELECTRICITY
  • H03—ELECTRONIC CIRCUITRY
  • H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
  • H03M7/00—Conversion of a code where information is represented by a given sequence or number of digits to a code where the same, similar or subset of information is represented by a different sequence or number of digits
  • H03M7/30—Compression; Expansion; Suppression of unnecessary data, e.g. redundancy reduction
• • A—HUMAN NECESSITIES
  • A61—MEDICAL OR VETERINARY SCIENCE; HYGIENE
  • A61B—DIAGNOSIS; SURGERY; IDENTIFICATION
  • A61B5/00—Measuring for diagnostic purposes; Identification of persons
  • A61B5/72—Signal processing specially adapted for physiological signals or for diagnostic purposes
  • A61B5/7232—Signal processing specially adapted for physiological signals or for diagnostic purposes involving compression of the physiological signal, e.g. to extend the signal recording period

Definitions

• OMFI analog-to-digital converted audio signal such as a music, electrocardiogram, or electroencephalogram signal
• OMFI analog-to-digital converted audio 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.
• 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 location.
• the encoded signal is decoded by a decoder, and the decoded signal is supplied to a digital decom ression 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 at zero-degrees, and the transfer function of the digital decompression filter
• OMPI includes poles on or inside the unit circle in the Z-plane at zero degrees. Compression filter operation is performed without truncation, or round-off, whereas decompression filter operation is with truncation, or round-off. In addition to the zero degree positions of the zeros the transfer function of the compression filter may also include zeros on the unit circle in the Z-plane at ⁇ 41.41", ⁇ 60°, ⁇ 90°, ⁇ 120° and /or 180°. An associated decompression filter is employed which has poles at, or inside, the unit circle adjacent the location of the compression filter zeros. The resultant system frequency response for analog signals is high-pass which may include one or more small high frequency notches.
• the resultant filter has a low frequency cut-off of between 0-15 Hz to accommodate audio frequency signals which range, appro imately, 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.
• 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 moving the poles of said filter inwardly, inside the unit circle thereby enabling the system to recover from said signal errors.
• error signal detection and inward movement of the poles of the transfer function in the Z-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.
• 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.
• transmission of actual signal values every, say, 6 to 16 milliseconds would be adequate.
• the number of successive actual signal values required 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.
• Figs. lA and IB together show a block diagram of a data reduction system; a digital recording and transmitter section being shown in Fig. 1A and a playback and receiver section being shown in Fig.
• OM ⁇ Fig. 2 shows a wavef orm and graphic representations of signals appearing at various locations in the data compression system shown in Figs. 1A and IB;
• Fig. 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 zeros 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 Huffman code of a type which may be used in the present invention.
• Fig. 10 shows the zero-pole pattern of a compression-reconstruction filter combination which 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 zero-pole pattern illustrated in Fig. 10;
• Figs. 12A and 12B are similar to Figs. 1A and IB, 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 c o m pr es si on -r ec o st r u ct io n f ilter combination which may be used in systems embodying the present invention.
• 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 periodically reinitialize the same.
• FIG. 1A 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
• an analog signal 22 is shown which comprises an input to the analog to digital converter 20.
• the audio input signal may comprise a music signal which ranges in frequency from appro imately 15 to 20,000 Hz.
• the form of the analog to digital converter output shown at B of Fig. 2, comprises samples f n -. j through f n+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.
• 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
• the output from the A/D converter 20 is supplied to a 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.
• OMPl receiver portion of the system is shown in Fig. 2A.
• the digital output from filter 23, as well as the digital input is identified as f n .
• an analog filter having a similar 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.
• the digital compression filter 30 is shown to include an estimator 32 and subtracting means 34.
• the estimator 32 provides an estimate of f n , here identified as f R , based upon actual samples occurring both before and after the sample f n to be estimated. Estimators for providing such estimated f n 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 f_ and the estimated signal value f n by subtraction of the estimated value f rom the actual value at subtracting means 34, as follows:
• 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 f n and an estimated value f n .
• Other compression filters may be used having different transforms in which the compression filter output ⁇ n is not a direct function of the difference between the actual input f and an estimated value thereof, f n .
• 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 copending U. S. Patent Application Serial No.
• the Huffman encoding technique makes use of the fact that the compression filter reduces the entropy of the signal output, ⁇ n so that there can be a reduction in the total number of bits in the Huffman encoded signal over the input signal.
• a single code word is assigned to infrequently occurring difference signals, and supplied as a label for the actual difference signal value n - In Fig.
• the encoder 40 output is designated h( ⁇ n ) and, at D in Fig. 2, the values h( ⁇ n ), h( ⁇ n+ ⁇ ) etc. represent encoded values of ⁇ n , ⁇ n+ ⁇ » 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 Application Serial No. 207,728 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 .
• the encoded value for ⁇ n+ 2 comprises a label together with the actual compressed signal ⁇ n+ 2 » wherein - ⁇ n+ comprises an infrequently occurring compressed signal value; that is, some value outside the range of ⁇ 3.
• the encoded signals from encoder 40 are recorded and/or transmitted to a remote receiver.
• 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.
• the encoder output is supplied to a buffer memory 52 and thence to a digital .modem 54 for transmission over transmission line 56.
• check bits are generated for recording and/or transmitting along with encoded compressed signals h( ⁇ n ).
• digital input signals f n sometimes are supplied to 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. 1A are reproduced using the system shown in Fig. IB, which system includes a playback unit 60.
• 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.
• the Huffman code words are converted to the original compressed signals ⁇ n .
• 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 copending U.S . Patent Application Serial No. 207,728. Coding and decoding are discussed in greater detail below under the heading "Encoding-Decoding”.
• the compressed signals ⁇ n from the decoder 66 are supplied to a reconstruc tion, or decompression, filter 70 through a buffer memory
• 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 f ilter 70.
• the reconstruction filter 70 converts the compressed signals ⁇ n to equal length sample signals f n (out) which closely match the input sample signals f_ to the compression filter 30.
• one feature of this invention involves compression filtering without truncation and decompression filtering with truncation.
• the truncated reconstruction filter output f n (out), f .. ⁇ (out) etc. is shown to comprise words of 24 bits.
• 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.
• the same symbol f n (out) is employed at the input and output of the filter 75.
• 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 thereof .
• the encoded signals are transmitted over line 56 (from Fig. 1A to Fig. IB) 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.
• a music analog signal is considered for an input to the present system of this invention.
• the size of the quantization level, q is small compared to the standard deviation, , of the analog music signal.
• the average word length of a Huffman encoder such as encoder 40 to which the output from the compression filter 30 is supplied is bounded as follows:
• the Z-transform of Eq (6) has two zeros at (1,0) and Eq (7) has two zeros at (l-2" m , 0) in the Z-plane.
• the ⁇ n in Eq (7) will have values spaced a distance of 2"" 2m q.
• the CTfrom both filters will be approximately equal but the ⁇ S" /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 .
• Any negative j and a non-zero b ⁇ • means that f n _ ⁇ is shifted right and added; j 0 bits must be added to the least-significant end of the arithmetic word, where j Q is the most negative j with non-zero b ⁇ •.
• 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.
• the frequency response of Eq(10) is:
• d is greater than one when • ⁇ ->60 ⁇ (f » 7.33KHz) so that spectral components above 7.33 KHz are amplified by d n for filters with all of the zeros at (1,0).
• n is a value of n such that increasing n above this value amplifies the total energy above 7.33 KHz more than the total energy below 7.33 KHz 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(Z) -K(l - z- l )
• a ⁇ are the constants that are used in Eq(8), and K is the gain constant.
• the a ⁇ are integers that can be expanded [as in Eq (9)] without negative j and, therefore, q is not reduced .
• a compression filter in which the Z-transform thereof has zeros at (1,0) and on the unit circle at at least one of the above- identified complex-pair positions (i.e. ⁇ 41.41°, ⁇ 60°, ⁇ 90°, ⁇ 120° and 180°).
• 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.
• 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 14 KHz) so that more zeros can be used at 1,0.
• 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 0°; 0° 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.
• the entropy that will be obtained can be estimated as follows: the music spectrum S(f) is measured, and the integral
• FIG. 8 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 which consecutive sample signals from the A/D converter, through filter 23, are shifted.
• the registers, 102, 104 and 106 are shown to contain samples f n , f n _ ⁇ » f n -2» respectively.
• 14-bit registers are employed for 14-bit samples.
• the register outputs are connected to a digital multiplexer 108 for selective connection of the sample signals to an arithmetic and logic unit (ALU) 110.
• ALU arithmetic and logic unit
• the digital compression filter 30 may include an estimator 32 having an output comprising an
• OMPI estimated sample value f_ based upon actual samples n _l and f n+ ⁇ occurring before and after the sample f n to be estimated.
• prior art estimators are used which provide an output
• equations (1) and (20) may be combined to give
• Equation (21) may be utilized by the illustrated compression filter in the generation of the compressed signal ⁇ n .
• An estimate f n of the sample f n is made using the samples either side of f n , i.e. f n _ ⁇ and f n+ t but not f n itself.
• the words f n _ ] _ and n + j are moved into the ALU 110 through the multiplexer 108 and added.
• the actual sample f n then is moved into the ALU 110 through the multiplexer 108, and multiplied by 2. Multiplying by 2 simply involves shifting of the bits toward the most significant bit.
• a table of compressed signals, __ n ranging from ⁇ 3 is shown together with a code word for said signals, the length of the code word, and the relative probability of occurrence of said compressed signals.
• the compressed signals ⁇ n which occur most frequently are assigned a code word.
• the probability of ⁇ n comprising a value which is assigned a code word is high, say, .98.
• 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.
• the reconstruction filter 70 transfer function 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.)
• one system includes the periodic transfer of a plurality of actual digital signal values, f n , 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 compression 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 n .
• poles of an exact inverse are off the unit circle if the compression zeros are not on the unit circle.
• 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 at a distance from the unit circle that is less than TT x 200/22000 (the distance around the unit circle that corresponds to 200 Hz). A larger distance means that the low frequency components will not be as highly attenuated. Since 0.00909 ⁇ » 2 "7 , the zero would be
• 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 to digital converter word length. Letting the output quantization level equal q , the quantizing noise po er will have a variance of
• 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 variance is
• g ⁇ is the ith value (at the ith sampling time) of the impulse response of the reconstruction filter.
• 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 .
• 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.
• the reconstruction filter 70 employed herein preferably comprises a digital computer programmed for the desired reconstruction filter operation.
• computers operating with such large word lengths are not practical for consumer music data compression.
• 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.
• noise generators with noise power q /12 are added to the filter input and the in put- to -o ut put standard deviation multiplier is calculated.
• the value of q is the quantization level of the truncated arithmetic word.
• 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 7 KH z on the unit circle, then a reconstruction filter having a complex pair of poles at the 20Hz Butterworth position and a complex pair 7.33 KHz along the unit circle and
• the standard deviation multiplier of such a reconstruction filter is 32, or 5 bits, and the
• ⁇ IPO frequency response of the compression- reconstruction filter combination is substantially the same as that shown in Fig. 11, except for a narrow notch at 7KHz.
• 24-bit arithmetic there is virtually no degradation of the signal.
• Digital computers with, say, 24 bit arithmetic for recons ruc ion filtering are available at a reasonable cost for use in the system of this inventio .
• 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.
• 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.
• 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.
• 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. 1A and is shown to include 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. 1A and described above, It will be noted that an analog high frequency deemphasis 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. 1A.
• 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.
• Fig. 12B 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 error is detected, a bit error signal is generated which signal is transmitted over line 94 and through switch 96 to the digital reconstruction filter for momentarily shifting the poles of the filter inwardly.
• a digital to analog converter 74 converts the signal samples f n (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. IB; i.e. to restore the amplitude of the high frequency signals which were deemphasized by filter 23A.
• 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 inside the unit circle, adjacent said zeros which provide for stable operation. Recovery of the reconstruction filt er from bi t 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.
• Fig. 13 wherein zeros and poles of the transf er f unction 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.
• this combination of zeros and poles is the same as that shown in Fig. 10, described above.
• the reconstruction filter poles are momentarily moved inwardly upon receipt of a bit error signal from the error checker 92 over line 94.
• the poles are shown moved to a point 0.0625 inside the unit circle for rapid recovery from the error.
• the poles of the reconstruction filter return to normal position, that is, to a point 0.00195 inside the unit circle at zero degrees.
• 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 y n determined by the ALU is loaded into register 140 while the prior value of y n is shifted from register 140 into register 142.
• the third register 148 is supplied with the sample value f n (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.
• the value stored in register 148 comprises f n _ ⁇ (out).
• error checker 92 When an error is detected by error checker 92, a smaller value of m 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 implementation of equations (25) and (26) simply by controlling the amount of shifting to perform the indicated multiplications by the factor 2 ⁇ m .
• contents of an ALU register are not shifted as far to the right for some 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 normal with the reconstruction filter poles again adjacent the unit circle.
• FIG. 12B shows a block diagram 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 which also are on the unit circle at the same locations as the zeros, during normal operation; that is during operation in the absence of bit errors.
• 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.
• the reconstruction filter 70A in the absence of transients, operates with poles on the unit circle, as shown 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 a point where m equals infinity.
• the reconstruction filter poles are momentarily moved inwardly, of the unit circle, at zero degrees.
• 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 f 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.
• the invention is not limited to inward movement of reconstruction filter poles to a single location in the presence of an error signal.
• m may be employed, with filter operation being stepped through several different pole locations during recovery from bit errors.
• 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.
• FIG. 1A and IB Apparatus illustrated in Figs. 1A and IB may be used for this type of operation.
• Huffman encoded difference signals h( ⁇ n+ ) h( ⁇ n+i ), Huffman encoded signal values h(f n ), h(f n+j ), etc. are periodically transmitted, by
• OMPI OMPI periodic actuation of switch 35 to the broken line position shown in Fig. 1A.
• switch 35 With switch 35 in the broken line position, a series of actual signal values, f are periodically supplied to the Huffman 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.
• 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 sig ⁇ nal value f n .
• 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 reconstruction filter 70.
• This periodic reinitialization of the reconstruction filter there is no requirement to operate the reconstruction filter wit-fc 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.

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• 1984
  • 1984-02-27 GB GB08519232A patent/GB2165426B/en not_active Expired
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