Method of and means for variable-rate coding of LPC parameters

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BACKGROUND OF THE INVENTION

This invention relates to linear predictive coding. In particular, it is a method and means of reducing the required channel bandwidth by minimizing the amount of information that is sent in transmitting an electrical signal representing speech or any other signal that has been subjected to linear predictive coding.

Linear predictive coding (LPC) is a method of digital coding that is particularly useful for voice signals. It makes use of the fact that in speech the pressure variations that constitute the sound follow patterns that stay the same for relatively long periods of time. LPC is normally applied to make use of four items of information about speech. The first of these is that speech may be either voiced or unvoiced. Voiced signals are signals that begin with a buzz from the vocal cords, while the vocal cords are inactive for unvoiced signals. Either voiced or unvoiced signals are further characterized by three sets of parameters. The first of these is energy or gain which is a measure of the loudness of a speaker. The second is pitch which is the fundamental frequency, if any, that is being generated by the vocal cords of a speaker. The third is some measure of the filtering effect of the vocal tract upon the vibrations generated by the vocal cords or other sound-generating mechanisms. Unvoiced signals are characterized only by energy and vocal-tract parameters; they have no pitch. The vocal tract is typically modeled as a stacked array of cylindrical tubes of varying lengths and diameters that has a series of damped mechanical resonances. The generation of a particular speech sound is carried out, both conceptually and actually, by passing the buzz of the vocal cords or the noise of moving air through the array of resonant structures. When a speaker changes the particular sound that he is making without changing his pitch, he is changing the dimensions of the resonant structures and hence the effect of the vocal tract upon the excitation signal generated by the vocal cords.

It would be possible to characterize the resonances of the vocal tract in a number of ways. These include characterizing the impulse response of the inverse filter, the coefficient values for the LPC direct-form filter, the autocorrelation function of the input speech, filter coefficients for the lattice filter (the so-called reflection coefficients k), the coefficients of the discrete Fourier transform of the autocorrelation function, and various transformations of the reflection coefficients. Speech can then be described in a digital system by developing digital characterizations of the voicing, the energy, the pitch, and of an equivalent filter. Because of the nature of speech, a particular set of filter coefficients will hold essentially the same values for tens or hundreds of milliseconds. This enables the characterization of speech to be made with sufficient fidelity by chopping that speech into frames of the order of 10 to 50 milliseconds in length and ignoring the possibility of any variation in the LPC parameters during that frame. It is also satisfactory in almost all instances to characterize the mechanical resonances by limiting the allowed number of reflection coefficients to ten.

Various transformations of the reflection coefficients have been used to describe the filter equivalent of the vocal tract. One that is of particular value is the logarithmic area ratio, abbreviated LAR which is the logarithm of the ratio of the magnitude of (1+k) to the magnitude of (1-k). A typical frame of speech that has been reduced to linear predictive coding will comprise a header to indicate the start of the frame and a number of binary digits during a period of time of the order of the frame period that signal voiced or unvoiced, energy level, pitch, and ten LARs. While the computation time necessary to determine the pitch and the LARs is of the order of the period of a frame or less, the systems of calculation that are available require information from a number of adjacent frames. For this reason, the information that is being sent to describe an LPC signal normally runs five, ten, or even more frames behind the speaker. Such a time delay is imperceptible to a listener so that the coding is properly described as taking place in real time.

A typical frame of speech that has been encoded in digital form using linear predictive coding will have a specified allocation of binary digits to describe the gain, the pitch and each of ten LARs. Each successive pair of LARs represents the effect upon the signal of adding an additional acoustic resonator to the filter. Limitation of the number of LARs to ten is in recognition of the fact that each additional reflection coefficient becomes progressively less significant than the preceding reflection coefficient and that ten LARs usually represent a satisfactory place to cut off the modeling without serious loss of response. The inclusion of more LARs would provide a marginal improvement in fidelity of response, but the difference between ten LARs and twelve seldom produces a detectable difference in the resulting speech. Furthermore, eight or even fewer LARs are often adequate for understanding. This makes it possible to use a system such as that of the present invention which uses redundancy to reduce the average bit rate and makes a temporary sacrifice of fidelity from time to time when it becomes necessary to reduce the bit rate below the average.

Systems for linear predictive coding that are presently in use have different frame periods and bit allocations. A typical such system is summarized in Table I which is a bit allocation for speech that was treated in frames 12.5 milliseconds in length. This corresponds to 80 frames per second. The voiced-unvoiced decision is encoded as one of the pitch levels so that a separate bit for voicing is not needed.

TABLE I ______________________________________ Bit Allocation for a Typical LPC System LPC Parameter Bits ______________________________________ Gain 5 Pitch 6 LAR 1 6 LAR 2 6 LAR 3 5 LAR 4 5 LAR 5 5 LAR 6 4 LAR 7 4 LAR 8 4 LAR 9 4 LAR 10 4 TOTAL 58 ______________________________________

Table I lists a total of 58 bits per frame which, for a frame width of 12.5 milliseconds, would represent a bit rate of 4640 bits per second. The addition of two more bits per frame that is necessary for synchronization raises the bit total to 60 and the bit rate to 4800 bits per second. The use of ten LARs in a frame length of 12.5 milliseconds gives excellent speaker recognition. It is desirable to retain that speaker recognition by retaining the same frame length and the same number of LARs and by keeping the same quantization of the LPC coefficients, all at a reduced bit rate.

One method of reducing the data rate for speech is to use a technique called Variable-Frame-Rate (VFR) coding. This technique has been described by E. Blackman, R. Viswanathan, and J. Makhoul in "Variable-to-Fixed Rate Conversion of Narrowband LPC Speech," published in the Proceedings of the 1977 IEEE International Conference on Acoustics, Speech, and Signal Processing. In VFR coding the first four LARs are examined every frame time. If none of the four LARs has a different quantized value, no information is transmitted for that frame. If any one of the four LARs has a change in quantized value from the last transmitted frame, then all of the parameters are transmitted for that frame. Hence, in this technique, all or none of the LPC coefficients are transmitted at each frame time. Since in some frames no data are transmitted, the resulting data rate is reduced. This has the disadvantage that if one LPC coefficient changes, all are sent, regardless of whether others may not have changed.

It is an object of the present invention to reduce the bandwidth necessary to send digital data.

It is a further object of the present invention to use redundancies of speech to reduce the bandwidth necessary to send a digital characterization of speech.

It is a further object of the present invention to transmit speech subjected to linear predictive coding in an average bandwidth that is less than the maximum bandwidth needed to transmit the encoded speech.

Other objects will become apparent in the course of a detailed description of the invention.

SUMMARY OF THE INVENTION

Digital data, and particularly speech that is encoded digitally by linear predictive coding, are compressed in bandwidth by storing data for each LPC coefficient for a predetermined number of frames. The quantized values of the coefficient for the first frame and that of the predetermined frame are used to determine a straight-line interpolation of the values of the coefficient at intermediate frames. The actual values of the coefficient at each intermediate frame are then compared with the interpolated values. If no value differs from its interpolated value by more than a predetermined threshold amount, only the last value is transmitted. If any value differs from its interpolated value by more than the threshold amount, a new straight-line interpolation is made using the quantized values of the coefficient for the first frame and one frame fewer than the predetermined value. This process is repeated if necessary until either a straight-line interpolation is achieved within the threshold or else the iteration has reached the next successive frame. Whenever data are sent, the entire process is repeated, taking as the first frame the last one sent. Received data are used to reconstruct coefficients according to the linear approximation. It should be evident that, if the threshold is equal to half the step size for quantization, this process introduces no more error than if the coefficients were quantized and transmitted every frame. The bandwidth of the data may be varied by varying the predetermined number of frames, the current number of LPC coefficients, the predetermined threshold, or a combination of these.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is an overall block diagram of a circuit for the practice of the present invention.

FIG. 2 is a block diagram of a portion of the variable-rate coding circuit of FIG. 1.

FIG. 3 is a block diagram of a counter circuit that generates timing waveforms for FIG. 2.

FIG. 4 is a flow chart of the operation of the circuit of FIG. 2.

FIG. 5 is a time plot of an interpolation.

FIG. 6 is a timing chart for FIG. 2.

FIG. 7 is a block diagram of a portion of the variable-to-fixed-rate coding circuit of FIG. 1.

FIG. 8 is a flow chart of the operation of the circuit of FIG. 7.

FIG. 9 is a timing chart for the circuit of FIG. 7.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT

FIG. 1 is an overall block diagram of a circuit for the practice of the present invention. In FIG. 1 an electrical analog of a waveform of speech or other signal exhibiting redundancy enters low-pass filter 10 which typically has a cut-off frequency of the order of 3 kHz in applications intended for voice radio communication. The filtered signal is applied to sampler 12 which is typically operated at a sampling rate of at least twice the cutoff frequency of low-pass filter 10. This produces as an output of sampler 12 a signal that contains the audio as pulse-amplitude modulation. The output of sampler 12 is applied to speech-parameter extractor 14 which uses any of a number of means, some of which are well known, to extract a set of parameters for linear predictive coding like those of Table I. While the description of the invention that follows will be given in terms of the bit allocation of Table I, it should be understood that different frame lengths, different coding schemes and different bit allocations could be handled equally as well.

The output of speech-parameter extractor 14 is a digital signal comprising synchronization bits, typically two bits, and 58 bits encoding the speech parameters associated with a particular frame. The synchronizing bits identify the beginning and end of an encoded frame. The transmitted frame containing 60 bits representing the synchronization bits and the LPC parameters is taken to parameter smoother 16 and then to variable rate coder 18. The parameter smoother is used to reduce small frame-to-frame variations in parameter values. This processing can be that of a low-pass filter, such as the weighted average of the parameter values over some number of frames, or it can be a median smoother whose output is the parameter value which corresponds to the median of all the parameter values over some number of frames. This smoothing operation, by reducing frame-to-frame variations, decreases the transmitted bandwidth by making a longer interpolation interval more likely.

Variable-rate coder 18 takes the digital data in a fixed frame length and processes that data in a way that will be described to produce frames of lengths that are less than or equal to the frame lengths of the input data. Variably encoded data from variable-rate coder 18 are applied to buffer 20 which restores, if necessary, a data rate that is less per frame than the fixed data rate that is the input to variable-rate coder 18. The digital output of buffer 20 is communicated over a channel 22 by any of a number of means such as modulation of an RF carrier, excitation of a telephone pair, or the like. Buffer 20 is also connected back to variable-rate coder 18 if desired to supply a control signal when a data rate approaches the capacity of channel 22. The control signal may cause variable-rate coder 18 to reduce the number of LARs transmitted. It may cause a change in an interpolating threshold to change sensitivity. It may change the number of frames over which interpolation is being performed. The output of channel 22 is applied to buffer 24 which, like buffer 20, has a storage capability to prevent the loss of data when changes are occurring rapidly. The output of buffer 24 is taken to variable-to-fixed coder 26 which recovers from the variably coded data an LPC signal that is again in the format of the bit allocation of Table I. This signal is applied to speech synthesizer 28 to recover speech which is a close copy of the input speech.

It is an observed feature of human speech that, when samples are taken in frames of the order of milliseconds or tens of milliseconds, there will often be little change in one or more of the LPC coefficients for a number of consecutive frames. This is true by inspection during the pauses between words and syllables, and it is nearly as obvious upon consideration of the number of times in speech that a particular sound is held for significant fractions of a second. The redundancy that is thus created provides a theoretical basis for the operation of the variable-rate coder 18 of FIG. 1. A common practice in LPC speech systems is to have a synthesizer change parameter values at a rate greater than only once per frame when a new value is received. The coefficient values are generally changed between one and four times a frame by forming a linear interpolation of the present frame value and the next frame value. This interpolated value is then used for part of the time of one frame before the next interpolated value is used. One reason for doing this is to prevent unduly rapid changes in coefficient values at frame boundaries. The present invention makes use of an interpolation process to reduce the amount of data that must be transmitted by extending the interpolation process over several frames.

In the present invention, it is necessary to transmit information describing over how many frames a parameter is to be interpolated as well as describing the parameter itself. One method is to send a number indicating which parameter is being sent, a number indicating how many frames have elapsed since the last time the parameter was transmitted and finally the value of the parameter. Another method would send a twelve-bit header every frame time which would indicated which of the twelve parameters (pitch, gain, and ten LARs) are transmitted this frame and then the values for these transmitted parameters. This method requires a twelve-bit header every frame. One method of reducing this overhead is to employ a hybrid technique which uses the present invention for the more significant parameters such as pitch, gain, and the first five LARs, and uses VFR for the less significant parameters, such as the last five LARs. This hybrid technique has been found to work satisfactorily and reduces the header requirements in this case from 12 to 8 bits per frame.

In the operation of the circuit of FIG. 1, the coefficient value for the frame last transmitted and the coefficient values for the next seven frames are considered together. If only this seventh frame is transmitted, six frames of data for that coefficient are not transmitted and the synthesizer uses coefficient values for these frames based on a linear interpolation of the quantized value for the last transmitted frame and the quantized value for the seventh frame. The differences between the true values and the interpolated value at each of those frames is error. If the error at each of these frames does not exceed a threshold, then the interpolation process is satisfactory and only the coefficient value corresponding to the seventh frame is sent. If the error at one of the intermediate frames exceeds the threshold, the linear interpolation is unsatisfactory. The process is then repeated by considering the transmission of the sixth frame and interpolating only over the intermediate five frames. This process is continued until the interpolation process does not introduce an error exceeding the threshold or until the process reaches frame 1, the frame adjacent to the last previously transmitted frame, in which case frame 1 is transmitted.

Because the exact coefficient value is not transmitted but only a code corresponding to a quantized value, some error is introduced even if a coefficient is transmitted. If the threshold value is chosen to be equal to the maximum quantization error, it is evident that this process of transmitting the coefficient introduces no more error than if the coefficient were coded and transmitted at every frame. In addition, this threshold can be made to vary to afford greater economies of transmission. One particular method which is part of this invention is to use a threshold value at every frame which depends upon the signal energy or upon the LPC gain term. It is known that vowel sounds are more significant to the perception of speech than are the unvoiced sounds. For this reason, the technique of gain suppression is employed which uses a smaller threshold value for frames that have high gain terms and larger thresholds for lower-gain frames. This threshold expansion for unvoiced sounds and particularly for silence allows for greater reduction of transmission.

A further refinement allows the threshold to vary with channel bandwidth. If the channel bandwidth becomes less than the average data rate from the variable-rate coder 18, the threshold values can be enlarged to cause a reduction in the amount of data transmission while allowing a gradual reduction in speech quality. This feature allows additional flexibility for systems where the channel bandwidth varies or the speech information from the LPC parameter extractor may vary, as might be the case between different speakers or with different languages.

FIG. 2 is a block diagram of a circuit realization of the variable-rate coder 18 of FIG. 1. FIG. 2 also includes speech-parameter extractor 14 and parameter smoother 16 of FIG. 1. In FIG. 2, sampled speech from sampler 12 of FIG. 1 enters speech-parameter extractor 14 which extracts LPC parameters and, if desired, smooths them. The smoothing operation of parameter smoother 16 is optional. The circuit of FIG. 2 would apply variable rate coding to just one of the LPC coefficients of Table I. To handle all twelve, it would be necessary either to have a parallel arrangement of twelve of the circuits of FIG. 2, one for each coefficient, or else it would be necessary to increase the number of shift registers to provide one of each of the shift registers for each LPC coefficient. Since the functions would be performed identically for each coefficient, the circuit of FIG. 2 will be described as it operates on one such coefficient which we choose here arbitrarily to be the pitch. The actual value of the pitch is taken either from speech-parameter extractor 14 or parameter smoother 16 on line 34. It is quantized in quantizer 36 and is encoded in binary form in encoder 38. The unquantized value on line 34 is taken as an input to multiplexer 40. The output of multiplexer 40 is taken as an input to shift register 42. The output of shift register 42 is cycled back through multiplexer 40 and into shift register 42 under the control of input MI. Shift register 42 is designed to handle a predetermined number of data elements of the pitch. We choose that number to be eight. The significance of that choice will become apparent later. Referring to Table I, pitch was there given a total of six bits. Thus, under the conditions just described, the actual digital value of the pitch will be represented in shift register 42 while the quantized value is stored in shift register 46.

The output of shift register 46 is taken back as an input to multiplexer 44 and is also taken as an input to subtractor 54 and multiplexer 56. The output of multiplexer 44 is also taken to latch 58, the output of which is applied to subtracter 54. The output of subtracter 54, representing the difference between the output of shift register 46 and the output of latch 58, is taken as an input to divider 60 which, in turn, feeds latch 62. The outputs of multiplexer 56 and latch 62 are subtracted in subtracter 64, the output of which is taken to latch 66. The output of latch 66 is taken back as an input to multiplexer 56 and is also applied to subtracter 68 where it is subtracted from the output of shift register 42. The output of subtracter 68 is compared in subtracter 70 with a threshold signal which is applied through OR gate 72 to D flip-flop 74.

The operation of the circuit of FIG. 2 will now be explained by referring to the counter array, FIG. 3, the flow chart, FIG. 4, the time plot of an interpolation FIG. 5, and the timing chart, FIG. 6 FIGS. 3, 4 and 5 indicate the source for various control and operating signals in FIG. 2. In FIG. 3, a counter 80 receives on terminal 82 a signal PC which is generated by speech-parameter extractor 14 of FIG. 1. The signal PC is generated as a positive pulse each time the particular LPC parameter in question is generated. Thus, it serves as a frame counter, and counter 80 contains the number of different parameters currently in the shift register. Counter 80 is connected to AND gate 84 to produce an output pulse that is taken to AND gate 86 when counter 1 is full. AND gate 86 also receives as an input the negation of PC to generate an output when counter 1 is full and PC goes low. The output of AND gate 86 is taken as an input to D flip-flop 88 which is clocked by a master clock pulse to produce timing signal MI. This is a pulse indicating that the shift register is full and that the process of interpolation should start.

The output MI of D flip-flop 88 is taken as a preset signal to counters 90 and 92. Counter 90 counts down to an A1 input signal which also shifts the parameter shift register so that its output is the proper parameter for interpolation. Counter 92 counts down on a timing signal A2 to hold the number of frames over which the parameter is to be interpolated. That number, C3, is taken as an input to counter 94 which is preset by an inverse of timing signal A2 and is caused to count down by timing signal A3. Counter 94 holds a number that is always less than or equal to the count C3 of counter 92. The count of counter 94 is the number of the frame that is being tested to see whether interpolation is allowed.

FIG. 4 is a flowchart showing the operation of the interpolation discriminator of FIG. 2. The flowchart of FIG. 4 carries out the process of interpolation that has been described. We again describe operation in terms of an interpolation over eight frames, although the number eight is a matter of design choice, and we suppose that the particular LPC coefficient being interpolated is the gain. On startup, operation box 100 transmits the code for the first gain coefficient. In operation box 102 the number of the first frame in the interpolation range or interval is set equal to one and the last frame in the interpolation range is M frames later, where M is typically seven. In operations box 104, the values of the gain for these eight frames are read in as X(J). These are the values stored in shift register 42 of FIG. 2. In operations box 106, the coefficient value for the last frame in the interpolation range is quantized and stored in Z(M). The quantized values for the eight frames are stored in shift register 46 of FIG. 2. In operations box 108, K is initialized to the second frame in the interpolation range, frame IST+1. N is set equal to the length of the interpolation interval, initially seven, and TT is the interpolation step size. The interpolation step size is the difference between the quantized values of the gain at the frames at the ends of the interpolation range divided by the number of frames in the interpolation range. The value of TT is held in latch 64 of FIG. 2. The interpolated value for the second frame in the interpolation range is computed and the difference between the interpolated value and the true value is found in operations box 110, which is the output of subtracter 68 of FIG. 2. This can be seen more clearly by referring to FIG. 5, which is a time plot of eight consecutive assumed values of a coefficient and interpolation over all eight. In FIG. 5, discrete times are given frame numbers ranging from first frame IST to last frame IEND. The endpoints of line 116 are the quantized values of X at frames IST and IEND. Line 116 is a linear interpolation of the values of coefficient X between frames IST and IEND. With values of K counted to the left from IST toward IEND, the equation of line 116 is given by

Z(K-IST)=Z(0)+TT*(K-IST)

where the points on the interpolation line 116 are X(K-IST) at frame K. Thus, referring to FIG. 5, point 118 is obtained by substituting the value of IST+1 for K in the expression for Z(K-IST). Point 120 is obtained by subtituting K=IST+2 and the other points follow.

Point 122 is the value actually obtained for the coefficient when K=IST+1. Point 124 is the value actually obtained for the coefficient when K=IST+2. The error between the interpolated value of point 118 and the actual value of point 122 is the length of line 126, which is given by the expression

X(K)-Z(K-IST)

when K=IST+1. Line 128 is the error when K=IST+2. Referring again to FIG. 4, operations box 110 calculates the first error, and decision box 132 tests whether that error is less than or equal to some threshold. Assume first that the error is less than the threshold. Operations box 134 then increases the value of K by one. Output is to decision box 136 which tests to see whether the interpolation process has reached as far as the last frame. If it has not, exit is by the "yes" line to re-enter operations box 110 with the new value of K. By this loop, all of the coefficients for frames between IST+1 and IEND-1 are tested to see of the error is less than or equal to the threshold. This process continues until decision box 136 indicates that the process has reached the last frame, at which point exit is from decision box 136 on the "no" line. This causes operations box 138 to transmit the code for the coefficient value for the last frame. Operations box 140 then updates so that the new first frame is the former last frame, and the new last frame is M frames later. In the circuit of FIG. 2, the code is held in latch 52 and counter 80 is preset to begin counting incoming frames until the next interpolation.

Suppose now that the error had exceeded the threshold at some point in the iteration of the loop represented by operation box 110, decision box 132, operations box 134 and decisions box 136. Exit is then to operations box 142. When the threshold is exceeded in an interpolation interval, the output of flip-flop 74 of FIG. 2 goes high. Since the threshold is exceeded, the attempted interpolation range was too great, and it will be necessary to try interpolation over the next smaller range. This is accomplished by reducing by one the number of the last frame in operations box 142. Decision box 144 then tests to see whether the number of the last frame is adjacent to the number of the first frame. If it is, interpolation is not possible and exit to operations box 138 directs transmission of the code for the coefficient of the current last frame. If reducing the number of the last frame by one does not place it adjacent to the first frame, then exit from decision box 144 is to operations box 146, which causes quantization of the value of the coefficient for the current last frame. This is taken as an input to operations box 108, which starts the interpolation process again over a number of frames that is one smaller than the last preceding interpolation.

The flowchart of FIG. 4 is a functional representation of the interpolatioon process of the circuit of FIG. 2, as plotted in FIG. 5. It is also an operating flowchart that can be used to construct a computer program for the realization of the interpretation process. Such a computer program is included here as Appendix A.

FIG. 6 is a timing chart for the circuits of FIGS. 2 and 3. FIG. 6 also includes the counters C1, C2, C3 and C4 on counters 1-4 respectively under the assumption that at this particular time interpolation over eight frames and over seven frames is not effective, but interpolation over six frames is effective. The various time waveforms and numbered elements of FIG. 6 are summarized in Table 2. Refer in FIG. 6 to line C1. This is a count of the number of parameters currently in shift registers 50, 46 and 42 of FIG. 2. When the number in counter 1 is less than seven, the shift registers are still filling and there is no information to be transmitted. This is true in the first time interval as evidenced by the fact that counter 1 lists six parameters in the registers for the first frame. Counters 2 and 3 are then in a don't-care state, denoted "x", and counter 4 has remained at zero from the last preceding interpolation. Action starts in the second time increment when counter 1 first counts to seven, indicating that the shift registers have data for the seven frames following the last transmitted frame as well as the data for the last transmitted frame. The full register coincides approximately with the end of a cycle of parameter calculation, and the end of that cycle is indicated by a pulse PC. This starts the interpolation process which continues during all of the time intervals when counter 1 is at seven. Counter 2 reflects the fact that all eight numbers in the shift register are read out for interpolation while they are simultaneously cycled back into the register. This occurs while counter 3 holds a count of seven. During this time counter 4 indicates that the interpolation process is testing each frame successively to see if the threshold is exceeded. Referring to waveform TE, it is evident that the threshold was exceeded in frame 4. This means that interpolation over eight frames is not satisfactory. Counter 3 is then incremented by waveform A2 to count down one number. Counter 4 now indicates that the outputs of frames 6 through zero are read out for interpolation. Waveform TE again indicates that the threshold was exceeded, this time at frame 1. The process is now repeated with counter 3 indicating interpolation over five frames. Since waveform TE has not gone high before counter 4 counts to zero, interpolation over five frames was successful. Waveform IC causes latch 1 to hold the value of frame 2 from counter 2 and, if desired, the interpolation interval of five from counter 3. This will be used together with the last value previously sent to interpolate and reconstruct data. Counter 1 is now set to two which is the number of parameters left in the shift register that need to be transmitted, and the process begins again as the registers are filled with five more parameters.

TABLE II ______________________________________ Waveforms and counter entries at FIG. 4 Identification function Descriptive Information ______________________________________ .0.A Clock Starts from PC high .0.B Clock Starts from PC high C1 Counter 1 Number of transmitted parameters in shift registers C2 Counter 2 Number of parameter at output of shift registers C3 Counter 3 Number of frames being interpolated over C4 Counter 4 Number of frame currently being tested PC Parameter complete Coefficients for frame have been calculated and are available A1 Timing Signal Causes interpolated value to be latched in operations box 66 and shifts out new coefficient value from shift registers. A2 Timing Signal Causes counter 92 to decrement to shorten interpolation interval A3 Timing Signal Causes TE to latch on clocks when valid differences are available from subtracter 70. A4 Timing Signal Causes latch 62 to hold inter- polation step size for current interpolation range. A5 Timing Signal Latch quantized value for last frame in interpolation range. A6 Timing Signal Controls multiplexer 56 to feed first quantized value of last transmitted frame to sub- tracter 64 to form first interpolated value and then to feed latch 66 output back to subtracter 64 to generate successive interpolated values. IC Interpolation Complete send coefficient code and frame number of last frame TE Threshold Exceeded interpolation range too long, try again over range one frame shorter. LAT high transition indicates interpolation complete and hardware ready to accept new input data. MI Multiplexer Input high causes shift registers to circulate outputs back to inputs. ______________________________________

One more pattern should be evident by inspection of the numbers on the counters 1 through 4 as shown in FIG. 6. If interpolation had not been possible over any interval, then counter 1 would have stayed at seven through the entire process of attempted interpolation. It would then have dropped to six for one frame and incremented to seven to start a new interpolation at the start of the next frame. Counter 2 would have counted from seven to zero eight consecutive times. Counter 3 would have continued the pattern evident there, counting from seven to zero. Counter 4 would have counted from seven to zero, six to zero, until finally it reached a count of one to zero indicating that interpolation was impossible and that the circuit should send the coefficient code for the next successive frame.

The result of the previous operations is to create a stream of digital data. Since the last value for one interpolation is the first value for the next interpolation, it need be sent only once, and the number of frames spanned can be determined by counting. If the LPC coefficient was changing so much at the time the interpolation process was started that interpolation is ineffective, then the transmission process may use more bits than would have been used without interpo