U.S. patent number 3,631,520 [Application Number 04/753,408] was granted by the patent office on 1971-12-28 for predictive coding of speech signals.
This patent grant is currently assigned to Bell Telephone Laboratories, Incorporated. Invention is credited to Bishnu S. Atal.
United States Patent |
3,631,520 |
Atal |
December 28, 1971 |
PREDICTIVE CODING OF SPEECH SIGNALS
Abstract
Predictive coding of signals, i.e., the reduction or redundancy
in a signal by subtracting from it that part which can be predicted
from its past, is a well-known technique for reducing the channel
capacity required to transmit a signal with specified fidelity. It
has been widely applied to signals, such as television signals
which have regularly repeating intervals of information, but has
not been satisfactorily applied to signals, such as speech, which
exhibit characteristics that vary from speaker to speaker and from
time to time for one speaker. According to this invention, an
adaptive predictor is employed which is readjusted periodically to
match the time-varying characteristics of a speech signal.
Inventors: |
Atal; Bishnu S. (Murray Hill,
NJ) |
Assignee: |
Bell Telephone Laboratories,
Incorporated (Murray Hill, NJ)
|
Family
ID: |
25030515 |
Appl.
No.: |
04/753,408 |
Filed: |
August 19, 1968 |
Current U.S.
Class: |
704/219; 375/250;
375/244 |
Current CPC
Class: |
H03M
3/022 (20130101); G10L 19/04 (20130101) |
Current International
Class: |
G10L
19/00 (20060101); G10L 19/04 (20060101); H03M
3/02 (20060101); G10l 001/06 () |
Field of
Search: |
;325/41,42,38.1
;179/15.55,1SA ;332/11D |
References Cited
[Referenced By]
U.S. Patent Documents
Other References
Linear and Adaptive Delta Modulation J. E. Abate, Proceedings of
the IEEE VOl. 55, No. 3, pages 298-308, March, 1967.
|
Primary Examiner: Claffy; Kathleen H.
Assistant Examiner: Leaheeny; Jon Bradford
Claims
What is claimed is:
1. Speech signal processing apparatus, which comprises:
means, adjusted in accordance with parameters representative of
identifying characteristics of selected pitch periods of an applied
speech signal, for predicting the present value of said speech
signal on the basis of signals in selected past intervals
thereof;
means for coding the differences between the predicted value and
the present value of said signal for transmission;
means for analyzing selected pitch periods of said speech signal to
develop a plurality of parameter signals which represent vocal
tract transmission and source characteristics of said speech signal
within said periods; and
means for periodically adjusting said predicting means in
accordance with said parameter signals.
2. Speech signal processing apparatus as defined in claim 1,
wherein,
said characteristics of said speech signal represented by said
parameter signals include the extent of selected past pitch periods
and the magnitude of signals within said pitch periods.
3. Speech signal processing apparatus as defined in claim 1,
wherein new parameter signals are developed every 5
milliseconds.
4. Speech signal processing apparatus as defined in claim 1,
wherein said means for predicting the present value of said applied
speech signal comprises,
a linear predictor characterized by a z-transform given by
where b is a factor representative of signal values during
consecutive selected signal intervals, K is a number representative
of the duration of consecutive pitch periods of said applied
signal, a.sub.m are amplitude factors representative of the short
time spectral envelope of said speech signal, and N represents a
selected number of said factors a.sub.m.
5. A communication system for conveying the information content of
a speech signal over a channel of relatively small capacity which
comprises, in combination:
at a transmitter station;
means for reducing the redundancy in a speech signal by subtracting
from it a predicted value of the signal derived from past pitch
period intervals thereof selected in response to parameter signals
developed from an analysis of selected pitch period intervals,
means for analyzing selected pitch periods of said speech signal to
develop a plurality of parameter signals which denote selected time
varying characteristics of said speech signal within said
intervals,
means for periodically adjusting said predicting means in
accordance with said parameter signals, and
means for transmitting both the difference between said predicted
value and the present value of a speech signal and said parameter
signals to a receiver station, and
at said receiver station;
means, adjusted in response to received parameter signals, for
predicting the value of said speech signal in response to
previously reconstructed speech signals, and
means for adding received difference signals to said predicted
value signals to develop a replica of said speech signal.
6. A communication system as defined in claim 5 in further
combination with,
means at said transmitter station for encoding said difference
signal and said parameter signals for transmission as a composite
signal, and
means at said receiver station for decoding said received signals
to recover said difference signals and said parameter signals.
7. A communication system as defined in claim 5 wherein said
difference signals and said parameter signals are transmitted to
said receiver station via diverse transmission facilities.
8. A communication system as defined in claim 5 wherein said
parameter signals are scrambled according to a prescribed code for
transmission.
9. Apparatus for predicting the present value of a speech signal
from its past, which comprises:
means supplied with reconstructed samples of a predictively coded
speech signal and with parameter signals which denote,
respectively, the values during each of a selected number of
consecutive intervals of said speech signal of the duration K of a
pitch period of said speech signal, the relative amplitudes b of
correlated signals in a number of said selected signal intervals,
and amplitude factors a.sub.m representative of the short time
spectral envelope of said speech signal during said selected
intervals, for developing signal samples that closely represent the
present value of said speech signal; and
means for periodically adjusting the values of b, K, and a.sub.m in
accordance with current speech signal values.
10. Apparatus for developing parameter signals for use in the
predictive coding of speech signals, which comprises, in
combination:
means for developing from past samples of an applied speech signal
a first signal which denotes the duration of a pitch period of said
applied speech signal;
means for developing from past samples of said applied signal a
second signal which specifies the relative amplitudes of correlated
signals in a number of selected consecutive intervals of said
applied speech signal;
means for developing from past samples of said applied signal a set
of signals which represents the short time spectral envelope of
said applied signal during said selected signal intervals; and
means for periodically selecting a number of consecutive intervals
of said applied speech signal to represent past samples of said
applied speech signal.
11. Apparatus for reconstructing a speech signal from signals
representative of the difference between the present value of said
speech signal and a predicted value derived from past pitch period
intervals thereof, which comprises,
means, adjusted in accordance with received parameter signals
representative of vocal tract transmission and source
characteristics of a speech signal, for predicting the value of
said speech signal in response to previously reconstructed speech
signals, and
means for adding said received difference signal to said predicted
value signal to develop a replica of said speech signal.
Description
BACKGROUND OF THE INVENTION
This invention relates to the efficient encoding of communication
signals and to the reduction of the channel capacity required to
transmit them. More particularly, it relates to the predictive
coding of speech signals. It has for its object a reduction of
redundancy in speech signals so that the signals may more
economically be transmitted to a receiver station.
1. Field of the Invention
The aim of efficient coding methods is to reduce the channel
capacity required to transmit a signal with specified fidelity. To
achieve this objective, it is often essential to reduce the
redundancy of the transmitted signal. One well-known procedure for
reducing the signal redundancy is known as predictive coding. In
predictive coding, redundancy is reduced by subtracting from the
signal that part which can be predicted from its past. For many
signals, the first order entropy of the difference signal is much
smaller than the first order entropy of the original signal; thus,
the difference signal is better suited to encoding for transmission
than the original signal. Predictive coding thus offers a practical
way of coding signals efficiently without requiring large capacity
storage facilities.
2. Discussion of the Prior Art
One of the principal methods for efficiently encoding communication
signals for transmission involves removing inherent signal
redundancy through the use of prediction apparatus at both the
transmitter and the receiver of a system. The current value of the
signal is estimated at both locations by linear prediction based on
the previously transmitted signals. The difference between this
estimate and the true value of the signal is quantized, coded and
transmitted to the receiver. At the receiver, the decoded
difference signal is added to the predicted signal to reproduce the
input speech signal. So long as a good prediction of the present
signal value can be made, efficient coding may take place. However,
speech is nonstationary so that a predictor with fixed coefficients
fails to predict the value of a speech signal efficiently. For
example, the speech signal is approximately periodic during voiced
portions; thus, a good prediction of the present value can be based
on the value of the signal exactly one period earlier. However, the
period varies with time so that the predictor must change with the
changing period of the input speech signal. Thus, since speech
signal statistics are not constant, it is necessary that the
prediction parameters be varied in accordance with the nature of
the incoming signal to adapt the predictor to the needs of the
signal.
SUMMARY OF THE INVENTION
In accordance with the present invention, redundancy in speech
signals is reduced by predicting the present value of the signal
from its past and by subtracting the predicted value from the
present value. To accommodate the constantly changing character of
speech, a form of predictive coding is employed in which the
predictor adapts to changing signal conditions. For speech signals,
past signal intervals are selected for prediction that are
comparable to individual pitch periods. The extent of the period
and the magnitude of the signal within the period are, in
accordance with the invention, periodically redefined. Preferably,
the parameter signals controlling the predictor are changed every 5
milliseconds. Such an interval has been found to be satisfactory
for accommodating the time-varying characteristics of the input
speech signal. The predictor parameter values are selected to
minimize the power in the difference signal averaged over 5-msec.
intervals. As such, the predictor parameter values constitute
slowly varying signals which can be transmitted efficiently.
To prevent errors that might be introduced in the system as a
result of the prediction and subsequent encoding processes from
being accumulated, it is in accordance with the invention to
reconstruct speech samples at the transmitter and to perform the
prediction operation on reconstructed speech signals and not on
input speech signals.
The difference between the present value signal and the predicted
value of the signal, if any, is eventually encoded and transmitted
to a receiver station together with the slowly varying parameter
signals which characterize the prediction.
Unlike previously proposed speech coding methods, the predictive
coding system of this invention accurately reproduces a speech
waveform rather than its spectrum. Listening tests show that there
is only slight, often imperceptible, degradation in the quality of
speech reproduced after transmission. In addition, experiments
indicate that binary difference signals and predictor parameter
signals prepared in accordance with the invention together can be
transmitted at rates of less than 10 kilobits per second, or
several times less than the rate required for ordinary pcm encoding
with comparable speech quality.
Since the difference signal developed in accordance with the
invention is the result of continuous efficient prediction, it has
low or zero intelligibility. It may be used, therefore, together
with the signals representative of the parameters of the adaptive
predictor, which are themselves unintelligible, to provide secure
transmission of speech signals. Only a recipient of the signals
with the appropriate decoding apparatus can properly reconstruct
the signals. Moreover, as a feature of the invention, the predictor
parameter signals may themselves be suitably scrambled for
transmission. Only if the appropriate key to scrambling is known
can they be properly recovered. Similarly, the difference signal
and the parameter signal may be transmitted via independent
channels as opposed to being multiplexed for transmission as a
composite signal.
DESCRIPTION OF THE DRAWINGS
The invention will be fully apprehended from the following detailed
description of a preferred illustrative embodiment thereof taken in
connection with the appended drawings in which:
FIG. 1 is a block schematic diagram of a transmitter station which
illustrates the principles of the invention;
FIG. 2 is a block schematic diagram of a receiver station
constructed in accordance with the principles of the invention;
FIG. 3 is a block schematic diagram of an adaptive predictor
suitable for use in the practice of the invention;
FIG. 4 is a block schematic diagram which illustrates the
construction of a suitable predictor parameter computer,
FIG. 5 is a block schematic diagram of a suitable arrangement for
computing the values of parameter .alpha. used for adjusting an
adaptive predictor used in the practice of the invention,
FIG. 6 is a block schematic diagram of a transmitter station in
accordance with the invention in which the difference signal and
the several parameter signals are conveyed to a receiver station by
separate transmission facilities, and
FIG. 7 is a block schematic diagram which illustrates the manner in
which the parameter signals may be scrambled for transmission.
DETAILED DESCRIPTION
A predictive coding system for speech signals in accordance with
the invention, includes: a transmitter (FIG. 1) for converting an
input speech signal to a low-bit rate digital signal for
transmission to a receiver; a predictor parameter computer (FIG. 4)
to calculate the parameters of an adaptive predictor (FIG. 3); and
a receiver (FIG. 2) to synthesize a speech signal from the received
digital signal.
Transmitter
A block diagram of a transmitter which illustrates the principles
of the predictive coding system of the invention is shown in FIG.
1. An input speech signal supplied at an input terminal is
initially filtered in conventional low-pass filter 10 and sampled
in sampling unit 11. In accordance with the well-known sampling
theorem, the sampling rate is twice the cutoff frequency of the
filter. A suitable sampling rate for speech signals is 6 kilohertz,
so that low-pass filter 10 has a cutoff frequency of 3 kHz. Speech
samples from sampler 11 are delayed by an interval of 60 samples
(10 msec.) by delay line 12 and delivered to one terminal of
differencing network 13, for example, a subtracting network. (Since
sampler 11 converts the input speech signal to a sequence of brief
samples, i.e., to digital form, it is appropriate to consider the
operation of the circuit on a sample-by-sample basis).
A predicted value Z.sub.N of the speech sample, obtained by
predicting (in network 30) the present value of the signal on the
basis, for example, of the value of past samples r.sub. N.sub.-1 ,
r.sub. N.sub.- 2,..., is delivered to a second terminal of network
13. The difference, .delta..sub.N, between S.sub.N and Z.sub.N, if
any, issuing from the network is next supplied to an adjustable
gain amplifier 14 and altered in amplitude by a factor Q. The
resultant signal is thereupon quantized to one of two levels, for
example, in two-level quantizer 15. To provide for the construction
of a predicted value of the signal, the signal developed by
quantizer 15 is altered in amplitude by a factor 1/Q in amplifier
16 and supplied to one terminal of adder network 17. The predicted
value Z.sub.N is supplied to the other terminal of adder 17. The
sum of these signals, designated r.sub. N, forms the reconstructed
signal. Note that at the Nth sampling instant, predictor 30 uses
only earlier samples (N-1, N-2,...) of the reconstructed signal.
The current sample r.sub. N of the reconstructed signal is formed
after the difference signal .delta..sub.N is quantized and added to
the predicted value Z.sub.N.
Adaptive predictor 30, which may be of the form illustrated in FIG.
3, is periodically adapted to accommodate changing signal
conditions, for example, in accordance with parameter signals
developed in computer 40. Details of a suitable computer are
discussed hereinafter with reference to FIG. 4. Predictor parameter
computer 40 operates on signal samples supplied directly from
sampler 11, and hence on signals in advance of their interaction in
the differential operation, since signal S.sub.N is delayed in unit
12 for a time sufficient to allow computer 40 to complete its
operations. With the form of computer employed in the illustrative
embodiment, it has been found that all computer operations can be
completed in the time required for approximately 60 samples;
signals supplied directly from sampler 11 thus are designated
S.sub. N.sub.+60. Parameter value signals, designated b, K and
.alpha., are thus developed to denote selected characteristics of
the speech signal on the basis of intervals corresponding nominally
to pitch periods of the signal. Parameter signal K represents the
duration of a pitch period of the applied signal, and parameter b
represents the relative amplitudes of corresponding signal values
in contiguous pitch periods used in the prediction operation.
Parameter signals .alpha. are amplitude factors related to the
formant structure of the vocal-tract transmission function and to
the spectral envelope of the vocal source. Similarly, computer 40
develops a signal, designated Q representative of the gain of
amplifiers 14 and 16 (and their counterpart at the receiver).
The binary signal at the output of quantizer 15, parameter signals
for adjusting the predictor, and the signal Q, representative of
the gain of amplifiers 14 and 16, thus constitute components of the
transmitted signal. They may be combined for transmission to a
distant station in any desired manner. For example, the binary
signal at the output of quantizer 15 may be supplied directly to
multiplexer 18 and the parameter value signals b, K and .alpha.,
and the signal Q may also be supplied to multiplexer 18 for
composite transmission to a receiver station. Alternatively, to
achieve a degree of signal security, the several signals may be
transmitted via independent channels as shown in FIG. 6. Moreover,
additional security may be achieved by scrambling the parameter
signals according to a known code prior to transmission. A suitable
arrangement is illustrated in FIG. 7. Scramblers suitable for
cryptically encoding signals are known to those in the art. It is
obvious that scrambled parameter signals and the difference signal
may be transmitted to a distant station in any desired fashion, for
example, by multiplexing as illustrated in FIG. 1 or by
transmission over diverse paths as shown in FIG. 6.
Receiver
A block diagram illustrating the various functions performed by a
receiver constructed in accordance with the invention is shown in
FIG. 2. Demultiplexer 21 serves to separate the various components
of the composite signal received at an input terminal, namely, the
quantized difference signal, signals denoting predictor parameters,
and a signal representative of the gain of the amplifiers used at
the transmitter. The predictor parameters are supplied to adaptive
predictor 30', which may be identical in all respects to adaptive
predictor 30 at the transmitter. The signal representative of the
gain Q is supplied to amplifier 22 and decoded difference signals
are delivered to amplifier 22. After being adjusted in gain by the
factor 1/Q, the difference signal is added to a predicted value
Z.sub.N ' of the present value of the signal developed at the
receiver, for example, in adder network 23. The reconstructed
samples r.sub.N ' are delivered to adaptive predictor 30' and also
supplied by way of low-pass filter 24 to an output terminal.
Low-pass filter 24, which has a cutoff frequency of one-half the
sampling rate, smooths the supplied samples to produce an output
speech signal r'(t). If there are no digital channel transmission
errors, evidently predicted values Z.sub.N ' are identical to
values Z.sub.N predicted at the transmitter, since predictor 30' is
adjusted identically to its counterpart 30 at the transmitter.
Hence, reconstructed sample r.sub. N ' is virtually identical to
r.sub. N at the transmitter. It is apparent that the error between
the reconstructed speech sample r.sub. N and the input speech
sample S.sub.N is identical to the difference .delta..sub.N "-
.delta..sub.N between the output of amplifier 16 and the input of
amplifier 14. Since, on the average, power of samples .delta..sub.N
is much smaller compared to power of samples S.sub.N, the
quantizing noise power in the reconstructed speech signal is a very
small fraction of the power in the input speech signal. Output
signal r(t) is thus an extremely close approximation to the signal
supplied as an input to the transmitter.
Adaptive Predictor
Two of the main causes of redundancy in speech are (1)
quasi-periodicity during voiced segments and (2) lack of flatness
of the short-time spectral envelopes. In accordance with the
invention, redundancy due to the quasi-periodic nature of speech is
reduced by a linear predictor consisting, for example, of a delay
and a gain. The z-transform of the predictor is given by
p.sub. 1 (z)=bz.sup..sup.- K (1)
where z.sup.- .sup.K represents a delay of K samples and b is an
amplitude factor. For voiced speech, delay K corresponds nominally
to a pitch period. The factor b compensates for possible unequal
amplitudes of the speech signal during contiguous pitch periods.
During the onset of voicing, b is frequently greater than unity;
the reverse is the case at the end of a voiced segment. For
unvoiced speech sounds, b is ordinarily close to zero.
Redundancy caused by the spectral envelope of speech is reduced, in
accordance with the invention, by means of an eighth-order linear
predictor. The z-transform of such a predictor is given by
An eighth-order linear predictor has been found to substantially
reduce redundancies due to three formants of the vocal-tract
transmission function and the spectral envelope of the vocal
source.
An adaptive predictor, which is suitable for speech signals, and is
in accordance with these considerations, is illustrated in FIG. 3.
It consists essentially of two separate linear predictor systems,
which exhibit transfer characteristics in accordance with equations
(1) and (2), and means for combining them.
Reconstructed signal samples, r.sub. n (delivered from adder
network 17 of the transmitter, and correspondingly, from adder
network 23 of the receiver, are delivered to storage unit 31. This
unit is equipped to store a variable digital signal y.sub. n for
values of n=-120, - 119,..., -1, 0, +1,..., + 29. Thus, it has a
storage capability of 130 digits. Of these, the last 30 digits are
replaced every 5 msec. Every 5 msec, storage unit 31 is actuated,
for example, by a pulse from clock 37, such that the signal in
storage location y.sub. 29 replaces the signal stored at location
y.sub.- .sub.1, the signal at y.sub. 28 replaces the signal at
y.sub.- .sub.2, and so on. Thus, every 5 msec. a new group of
samples is advanced into locations y.sub.- .sub.1,..., y.sub.-
.sub.120 to constitute a stored sequence of "past" samples. The
locations, y.sub. 29,..., y.sub. 0, are vacated and made available
to incoming reconstructed value signals r.sub. N for the next 30
sample intervals.
During each 5 msec. interval, the values of y.sub. n stored in
locations y.sub.- .sub.1,..., y.sub.- .sub.120 are delivered
sequentially to arithmetic unit 32 which is equipped to compute
value, c.sub. 1, in accordance with equation (1 a) for each value
of y.sub. n. Equation (1 a), viz,
c.sub. 1 = by.sub. n.sub.- K (1 a)
defines an input-output characteristic which corresponds to the
form of equation (1), and specifies an output signal c.sub.1 for
each supplied value of y.sub. n.sub.-k. The necessary factors, b
and K, are supplied to arithmetic unit 32 from predictor parameters
computer 40 (FIG. 1). The resulting signals are delivered both to
arithmetic unit 33 and to arithmetic unit 34.
Arithmetic unit 33 is programmed to develop values of u.sub.n in
accordance with the relation
u.sub. n =r.sub. N -c.sub. 1. (3)
The momentary value of r.sub.N is supplied to arithmetic unit 33
from the input to adaptive predictor 30. Evidently, arithmetic unit
33 comprises a simple subtractor network.
Similarly, storage unit 35 provides digital storage facility for a
variable u.sub.n for values of n=-8,..., -1, 0, +29. Unit 35 may be
a shift register or the like. It is reset every 5 msec. by a pulse,
for example, from clock 37, to shift the signals stored in
locations 22 through + 29 into the first eight locations and to
free the locations 0 through +29 for incoming signals. The vacant
storage locations are filled progressively with values of signal
u.sub.n developed by arithmetic unit 33. During each 5 msec.
interval, values of u.sub.n stored in unit 35 are delivered to
arithmetic unit 36 which is arranged to compute values of c.sub.2
in accordance with equation (2a), as follows:
Equation (2a) corresponds to the generalized relation of equation
(2). In essence, arithmetic unit 36 is a cumulative multiplier
network which sums the product of .alpha. and u for values of m=1
through m=8 for each value of u.sub.n supplied from storage. The
necessary amplitude factors .alpha. are supplied to unit 36 from
predictor parameters computer 40 (FIG. 1). Computed values of
c.sub.2 are delivered to arithmetic unit 34 wherein they are
arithmetically added to values of c.sub.1 supplied by arithmetic
unit 32 in accordance with equation (4) as follows:
Z.sub. N =c.sub. 1 + c.sub.2. (4)
Evidently, arithmetic unit 34 may comprise an adder network.
The resulting values of Z.sub.N constitute the predicted value of
the incoming speech signal sample S.sub.N and are delivered, as an
output signal, to subtractor network 13 of the transmitter (FIG. 1)
and, correspondingly, to adder network 23 of the receiver (FIG. 2).
The above-described arithmetic operations are carried on
sequentially for each value of n from 0 through 29. In the above
discussion, the integer N indicates the count of the current sample
of the input signal, i.e., from sampler 12 (FIG. 1), minus 60
samples to take account of the 10 msec. delay. The integer N
indicates a corresponding count within each unit. Variables u.sub.n
and r.sub.N are consecutively stored in storage units 81 and 35,
respectively, in locations 0 through 29. Every 5 msec., both
storage units are reset, as described above, and consecutive
samples of r.sub.N are again stored in locations 0 through 29 in
storage unit 31, and consecutive samples u.sub.n are again stored
in locations 0 through 29 and storage unit 35.
Predictor Parameters Computer
Parameters for the adaptive predictors at the transmitter and
receiver stations are calculated in special computation apparatus
which may be of the form illustrated in FIG. 4. Such apparatus
develops the predictor parameters necessary to adjust the predictor
optimally despite the nonstationary, time-varying character of the
input speech signals. Predictor parameters are recalculated every 5
msec. to ensure that prediction is efficient even when the speech
characteristics are changing relatively fast.
Input speech samples, S.sub.N.sub.+ 60, from sampler 11 (FIG. 1)
are supplied to storage unit 41 which is equipped with sufficient
storage capacity to accommodate an array w.sub. n in a
configuration identical to that described above. Incoming samples
are thus stored in the array as w.sub.-.sub.120,..., w.sub.-.sub.1,
w.sub. 0,..., w.sub.29. The sample at location w.sub.0
=S.sub.M.sub.+60, that at location w.sub.1 =S.sub. M.sub.+61 ,...,
and so on through w.sub.29 =S.sub. M.sub.+89, where M indicates the
sample number of the first sample of the current "frame" of
samples, i.e., samples in a 5 msec. group. Storage unit 41 is reset
every 5 msec., for example, by a pulse from clock 37 of FIG. 3
(connections not shown for simplicity) such that w.sub.j
=w.sub.j.sub.+30 for all values of j=-120,..., -1. Accordingly,
storage locations w.sub.0,..., w.sub.29 are vacated every 5 msec.
and used to store the new samples incoming from sampler 11. The set
of 30 newly installed samples constitutes a new frame of
signals.
Signals from storage unit 41 are supplied in parallel to arithmetic
unit 42 wherein computational values .chi..sub.j according to
equation (5), are computed as follows:
Arithmetic unit 42 includes individual computational units, 42a,
42b ,..., 42n, which operate in parallel to compute .chi..sub.j
according to the equation for values of j=15 ,..., 120. A special
purpose computer programmed according to the equation to be
employed to evaluate these signal values or, alternatively, several
individual arithmetic operations, e.g., multiplication, summation,
rooting, and division, may be performed serially according to
techniques well known in the art.
The computed array of values of .chi., i.e., .chi..sub.15,...,
.chi..sub.120,..., are supplied in parallel to peak locating
network 43 wherein the largest value of .chi. is determined. Thus,
peak locating network 43 finds the value of j such that .chi..sub.j
is the maximum of all values of .chi.. Networks for picking the
"biggest" from among a plurality of signals are well known in the
art; a suitable one typically includes a progressively biased diode
matrix. The index of the largest selected value of .chi. is
designated K and is supplied as one parameter necessary to adjust
the adaptive predictors at the transmitter and receiver
locations.
Parameter K is also supplied to B computer 44 and to arithmetic
unit 45. Computer 44 is also supplied with the signals w.sub.n from
storage unit 41. It computes b from these data according to
equation (6) as follows:
As before, a special purpose computer or a conventional arrangement
of arithmetic units may be employed to evaluate b. The computed
value of b similarly constitutes a parameter necessary to adjusting
the adaptive predictors. Values of b are also supplied to
arithmetic unit 45.
Arithmetic unit 45 is scheduled to develop an array of signal
values u.sub.n according to equation (7), below, for values of
n=0,..., 29.
u.sub.n =w.sub.n -bw.sub.n.sub.-j
n=0,..., 29. (7)
Values of signals in the array w.sub.n are supplied to arithmetic
unit 45 from storage network 41.
The various computations outlined above are carried out serially in
the order stated. The suboperations, e.g., the computation of
values of .chi. in arithmetic unit 42, b in computer 44, and
u.sub.n in arithmetic unit 45, are carried out in parallel circuits
within those units.
Every 5 msec. the array of signal values u.sub.n is transferred
into storage unit 46 to replace the previous arrays of signals in
storage. Storage unit 46 thus stores an array of signal values
u.sub..sub.-8, u.sub..sub.-7,..., v.sub..sub.-1, u.sub.0,...,
u.sub.29. Every 5 msec. the values u.sub.- .sub.8,..., u.sub.-
.sub.1 are replaced by the valves u.sub.22,..., u.sub.29. The
incoming samples are placed in the vacated storage locations
u.sub.0,..., u.sub.29. Thus, the signals u.sub.0,..., u.sub.29, are
consecutively stored as they are received in storage unit 46.
Periodically, under the influence of clock signals, an array of
signal values u.sub.n are read out of storage unit 46 and
transferred to arithmetic unit 47A. This unit comprises 36
arithmetic units designated f.sub.1,1 ; f.sub.1,2,..., f.sub.1,8 ;
f.sub.2,2 ; f.sub.2,3,..., f.sub.2,8 ; f.sub.3,3,..., f.sub.8,8,
which operate in parallel. Each unit serves to compute one value of
f according to equation (8) as follows:
Computations of f.sub.i,j are carried on simultaneously and the
output values, designated F, are periodically supplied to computer
48.
The array of signals u.sub.n is also supplied to arithmetic unit
47B wherein an array of values are evaluated according to equation
(9) as follows:
Arithmetic unit 47B preferably comprises an array of eight
individual units operating in parallel to evaluate the several
values of g. The resultant array, g.sub.1,..., g.sub.8, designated
G, is delivered every 5 msec. to computer 48.
Computer 48 is programmed to solve the matrix equation
F.alpha.=G (10)
to yield values of .alpha.. Although any special purpose computer
may be programmed for this evaluation, one suitable arrangement is
described below with reference to FIG. 5. Suffice it to say at this
point that the output of computer 48 is an array .alpha. of signal
values a.sub.1, a.sub.2 ,..., a.sub.8, which constitutes parameter
values necessary for adjusting the adaptive predictors at the
transmitter and receiver stations. These signals are thus applied
directly to adaptive predictor 30 at the transmitter (FIG. 1), and
to multiplexer 18 at the transmitter for delivery to the receiver
and adaptive predictor 30'.
Array .alpha. is also delivered to Q computer 49. Computer 49
constitutes an arithmetic unit arranged to evaluate values of Q
according to the relation
Arithmetic units for obtaining products, summations, differentials,
absolute values and so on, are well known to those skilled in the
art. Values of Q thus evaluated are used both at the transmitter
and at the receiver to set the gains of the several adjustable gain
amplifiers used in the predictive networks. At the transmitter,
values of signal Qare used to set the gains of amplifiers 14 and
16; at the receiver to set the gain of amplifier 22.
Although the several individual processing steps required to
evaluate the various intermediate parameter values take place
sequentially in the apparatus of FIG. 4, it is evident that
essentially instantaneous processing takes place in the various
computational units within each frame interval. The various sets of
parameter signals are advanced, one unit to the next, for example,
in accordance with pulses from a clock (such as clock 37 in FIG.
3).
The various predictor parameters and gain factor Q are recalculated
every 5 msec. These calculated values are held fixed for a duration
of 5 msec., the period over which the predictor parameters have
been optimized. Due to the 10 msec. delay of incoming signals at
the transmitter, the predictor parameters computer calculates the
parameters ahead of the time they are needed at the transmitter.
The adaptive predictors are reset just before the arrival of the
first speech sample of each frame at the transmitter.
Parameter .alpha. Computer
Operations sufficient for evaluating .alpha. in accordance with
equation (10) are described, for example, at pages 145-146 of
"Computational Methods of Linear Algebra" by D. K. Feddeev and V.
N. Faddena (English translation by R. C. Williams published by W.
H. Freeman & Co., San Francisco, 1963. Although conventional
operations as described in the literature may be employed, one
arrangement that has been found particularly suitable is
illustrated in the block schematic diagram of FIG. 5.
In FIG. 5, the array of signals F, representative of values of f
developed in arithmetic unit 47A, are supplied, respectively, to
arithmetic units 51. The system of arithmetic units operates on the
supplied values of f to produce an array of modified functions,
designated h.sub.i,j for values of i=1 ,..., 8, and for values of
j= i,..., 8. Values of h are individually stored in storage unit
52. Arithmetic unit 51a, for example, develops a value of h.sub.1,1
in accordance with equation (12), as follows:
h.sub.1,1 = f.sub.1,1 (12) Apparently arithmetic unit 51a comprises
a square rooting device. Values of h.sub.1,2,..., h.sub.1,8, are
evaluated in arithmetic unit 51b in accordance with the relation
shown in equation (13), viz,
h.sub.1,j =f.sub.1,j /h.sub.1,1, j=1 ,..., 8. (13)
Evidently, arithmetic unit 51b comprises a plurality of individual
units for developing a quotient signal. The necessary value of
h.sub.1,1 is delivered to arithmetic unit 51b from storage unit
52.
In like manner, values of h.sub.i,j are sequentially derived in
arithmetic units 51, progressing from left to right in the drawing,
according to the relation: ##SPC1##
Evidently, units shown in the drawing and designated 51c, 51e, 51g,
make the evaluations according to equation (14), i.e., for values
of h.sub.2,2, h.sub.3,3,..., h.sub.8,8. The remaining evaluations,
according to equation (15), are made in units 51d, 51f, and so
on.
It is, of course, possible that values of h.sub.i,j are at times
zero. Hence, to avoid any ambiguity in evaluating functions
according to equations (14)and (15), it is in accordance with the
invention to prescribe an arbitrary rule to accommodate this
situation. Any similar rule may, of course, be used. According to
the selected rule an increment .epsilon. is added to each input for
a detected input of zero. As a result, the input always assumes a
finite value and the required division operation can take place. A
small signal .epsilon., derived for example from battery 53, is
supplied to adders 54 in the input circuits of the requisite
arithmetic units 51. The magnitude of .epsilon. is selected in
accordance with the relative signal magnitudes accommodated by
units 51, to be insignificant as far as signal evaluation is
concerned, but sufficient to avoid the divide-by-zero ambiguity. If
desired, switch 55 may be used to open the .epsilon. circuit except
when zero signal is detected.
The computed values of h are supplied to arithmetic units 56
together with values of G (from arithmetic unit 47B) and functions
p.sub.i are developed as follows: ##SPC2## The array of values of
p, viz, p.sub.1,..., p.sub.8, is stored in storage apparatus 57 and
supplied as required to arithmetic units 58 wherein an array of
signal values .alpha. is developed for values of i=1 ,..., 8
according to the equation ##SPC3## The necessary values of h for
this evaluation are supplied from storage unit 52. The resulting
array of values .alpha. is delivered to storage apparatus 59.
Periodically this array is delivered to arithmetic unit 49 for the
evaluation of Q and to the adaptive predictors 30 and 30' (FIG.
3).
It is apparent that the apparatus described herein represents
merely one suitable manner of carrying out the necessary operations
to adaptively predict the values of a speech signal to promote
efficient coding for transmission. Numerous alternative techniques
may be employed for the evaluation; in fact, many of the operations
may be programmed for evaluation by a special purpose computer.
Moreover, the signals prepared for transmission may be combined in
any desired fashion or, in the alternative, may be transmitted
separately to achieve secure transmission of the speech
signals.
It is of interest that the quantizing noise appearing at the output
of the receiver, as described in this invention, is essentially
white in nature (flat spectrum). Frequently it is desirable that
quantizing noise have a nonflat spectrum. For example, noise whose
spectrum is weighted down at high frequencies may be subjectively
less annoying. Any desired noise spectral characteristics can be
obtained by employing a suitable preemphasis network before
low-pass filter 10 in the transmitter and a deemphasis network
after low-pass filter 24 at the receiver. A suitable preemphasis
characteristic for speech signals is one which is flat up to about
500 Hz. and rises at about 10 db. per octave between 500 and 300
Hz. It is not necessary that a preemphasis network be used prior to
low-pass filtering. It may, for example, be used just after the
sampler 11. Similarly, the deemphasis network may be used just
prior to low-pass filter 24 in the receiver.
In all events, the above-described arrangements are merely
illustrative of the application of the principles of the invention.
Numerous arrangements may be devised by those skilled in the art
without, however, departing from the spirit and scope of the
invention.
* * * * *