U.S. patent number 3,870,817 [Application Number 05/368,264] was granted by the patent office on 1975-03-11 for phonetic sound recognizer for all voices.
Invention is credited to Meguer V. Kalfaian.
United States Patent |
3,870,817 |
Kalfaian |
March 11, 1975 |
Phonetic sound recognizer for all voices
Abstract
In an articulated phonetic sound wherein the information bearing
group of format resonances varies within different frequency
regions in the sound spectrum, such as a phonome by
diffferently-pitched speakers the filter-separated signals derived
from said group of resonances are regrouped (shifted) in
prearranged combinations sequentially until a reference regrouping
is established for neutralizing (normalizing) the undesired effect
of said variations for adaptation to standard analysis. A specific
group of signals from said regrouped signals are then selected and
their amplitude ratios one with respect to another are further
matched with standard groups of amplitude-ratio measuring means for
obtaining a null output representing a good amplitude match.
Inventors: |
Kalfaian; Meguer V. (Los
Angeles, CA) |
Family
ID: |
26904366 |
Appl.
No.: |
05/368,264 |
Filed: |
June 8, 1973 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
Issue Date |
|
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209661 |
Dec 20, 1971 |
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Current U.S.
Class: |
704/235 |
Current CPC
Class: |
G10L
15/00 (20130101) |
Current International
Class: |
G10L
15/00 (20060101); G01l 001/00 () |
Field of
Search: |
;179/1SA,1SB,15.55R,15.55T |
References Cited
[Referenced By]
U.S. Patent Documents
Primary Examiner: Claffy; Kathleen H.
Assistant Examiner: Kemeny; E. S.
Parent Case Text
This is a continuation of application Ser. No. 209,661, filed Dec.
20, 1971, now abandoned.
Claims
What I claim, is:
1. In complex sound waves wherein the information is contained
within the time periods of trains of waves, and said information is
represented by the frequency and amplitude ratios between a group
of resonances in each train of the complex wave, but wherein
spectral variations occur, for different voices, the system for
detecting and storing the peak amplitudes of said group of
resonances during the time periods of said trains successively,
adaptable to stored-signal regrouping as a representation of
normalyzing said spectral variations, and analyzing the amplitude
ratios between said regrouped signals, said system comprising a
plurality of band-pass filters to separate the voice signal's peak
amplitude resonances into corresponding channels, each filter
output feeding a parallel-pair of channels of
detector/gate/storage/ discharge and mixer-gate channels to a
common gated mixer output, said parallel pair of detector channels
for alternate switching and storage of the filter output, where a
plurality of detector means for detecting and storing the peak
ampltudes of the resonances in said complex wave, each detector
means consisting of a first pair of storage means for alternate
storage of said detected signals, a second pair of discharging
means for said storages, and a third pair of mixing means for said
stored signals to a common output terminal; means for producing
undirectional timing pulses at the frequency rate of said trains; a
flip-flop (block 20), and means for operating it by said
unidirectional timing pulses for producing alternate steady state
output signals at the rate of said trains; coupling means from said
steady state signals to said first pair of storage means
alternately for effecting alternate signal storages of the detected
peaks of said resonances during the successive time periods of said
trains; normally inoperative A and B AND-gates (blocks 21, 22);
parallel connections of the first inputs of said A and B AND-gates;
direct coupling means from said steady state signals alternately to
the second inputs of said A and B AND-gates; a one-shot (block 35)
having operating time period not longer than the shortest time
period that occurs during said trains; coupling means from said
one-shot to said parallel connected first inputs of said A and B
AND-gates; coupling means from said unidirectional timing pulses to
said one-shot for operation, whereby operating said A and B
AND-gates alternately during the first time portions of said
trains; means for deriving alternate timing pulses (from block 20)
at the frequency rate of said trains; first, second and third,
fourth set-reset flip-flops (R-S blocks 23, 24 and 25, 26), each
flip-flop having set and reset inputs and set and reset outputs,
respectively; A parallel coupling means of the set input of the
first flip-flop with that of the reset input of said fourth
flip-flop, and B parallel coupling means of the set input of the
second flip-flop with that of the reset input of the third
flip-flop; means for applying said alternate timing pulses to said
A and B coupling means alternately, whereby effecting operation of
the first and second flip-flops in alternate set states during the
operating time periods of said one-shot, and operation of the third
and fourth flip-flops in alternate reset states during quiescence
of said one-shot; and coupling means from the set outputs of said
first and second flip-flops to said third pair of mixing means, and
the reset outputs of said third and fourth flip-flops to said
second pair of discharging means, for effecting storages of the
detected signals of said common outputs, adaptable to said spectral
normalization and analysis of said amplitude ratios.
2. The system as set forth in claim 1, wherein is included coupling
means from said common outputs to a matrix of signal regrouping
combinations, each combination having been prearranged for a
specific regrouping of the signals at said common outputs under
control of a control signal derived from a stored signal appearing
at a specific output of said common outputs; coupling means from
said common outputs to a plurality of signal sensing means,
respectively, for deriving sensing signals from the stored signals
at said common outputs; coupling means from said sensing means to
the first inputs of a plurality of control AND-gates, respectively;
means for applying sequential pulses to the second inputs of said
plurality of AND-gates, whereby only those AND-gates which have
received sensing signals at their first inputs operate by the
distribution pulses, for producing said control signals; and
coupling means from said control signals to said matrix for
operating the respective signal regrouping combinations of said
matrix, for regrouping the stored signals at said common outputs as
a representation of said spectral normalization.
3. The system as covered by the claims 2, which in combination
include a plurality of prearranged combinations of signal matching
means; coupling means from said regrouped stored signals to said
plurality of signal matching means for deriving a null signal from
the combination that a match has been obtained by one of said
sequential pulses, as a representation of said amplitude ratio
analysis.
Description
This invention relates to phonetic sound wave analysis, and more
particularly to an improved and composite arrangement of my
previously proposed systems for contructing a practical automata
capable of recognizing phonetic sounds as spoken by different
speakers.
One of the important contributions in the present invention is the
improvement over my switching arrangement as disclosed in my
related U.S. Pat. No. 3,622,706 issued Nov. 23, 1971, and related
U.S. Pat. No. 3,678,201 issued July, 18, 1972 which I had devised
for modifying the characteristic components of the sound wave in
such simulated manner that the spectrum variations which normally
occur in oral speech are neutralized (normalized) during the
analytical process of recognition.
One of the most problematic conditions in speech sound wave
analysis is the spectrum variations that occur not only in
different speakers, but also in the voice of a single speaker.
Changes in pitch frequency change the absolute values of the
frequency peak resonances (or formants as they are called in speech
signal processing), also harmonic ratios and amplitude ratios are
maintained for the same speech sound (for example, a specific
phoneme). Systems have been previously proposed to neutralize these
variations, and the practice has generally been called frequency
conversion; frequency normalization; and frequency standardization.
To be specific in the terminology used herein accordingly, the term
"normalization" will be used hereinafter throughout the
specification and claims. These systems, however, have one way or
another been subject to critical controls, and the practice has not
been favored for acceptance in experimental systems for speech
recognition automata. In carrying out an exemplary method of
spectrum normalization, the group of resonances in the arriving
wave are regrouped in a reference frequency region in the sound
spectrum in an order that, the lowest frequency (pitch) in the
group is converted to a reference pitch frequency, and the other
frequencies are converted to frequencies by the same factors of
multiplication from the reference frequency as they differ from the
original pitch frequency. Since in transmitting techniques of
information, as known and practiced, one set of parameters may be
substituted for another without loss of definition as long as the
independent parameters remain unchanged, it is readily seen that
the converted frequencies are relocated in fixed (standard)
positions in the sound spectrum, and therefore, easily adapted to
any type of analytical processing that may be desired. For
practical purposes, however, the method just mentioned involves
critical adjustments. Accordingly, the novel switching system used
herein may be briefly described, as in the following:
There are used a bank of channels arranged in numerical order, such
as 1, 2, 3 . . . n each one of which is provided with a plurality
of signal-admitting inputs, and a plurality of signal-switching
inputs, respectively. The detected signals derived from various
resonances (sub-band divisions in the sound) are applied to the
plurality of signal-admitting inputs, so that anyone of the signals
can be admitted to the output of anyone of the channels by the
operation of a respective signal-switching input. Thus in order to
obtain neutralization of the frequency variations, a plurality of
prearranged combinations of groups of switching signals are applied
to the plurality of signal-switching inputs for regrouping
(shifting) the detected signals admitted to the outputs of the bank
of channels sequentially until a specific group of the detected
signals are regrouped along a reference standard region of the
numerically arranged channel outputs. A group of signals from the
regrouped signals at the channel outputs are then selected and
their amplitude ratios one with respect to another are matched with
plurality of prearranged groups of amplitude-ratio measuring means,
for deriving from one of the last said means a final signal
representative of the originally spoken phonetic sound.
As explained in my reference patents, the time period devoted to
performing all of said combinations of signal regroupings is
performed within a portion of a pitch period, which may not be long
enough to render the system versatile for various other relative
uses. In order to reduce the number of signal regrouping operations
during each pitch period, we may first refer to the lowest (pitch)
frequency in a group of resonances representing a phonetic sound,
which vary from 45 to 600 cycles per second, or higher, if singing
is also to be considered. In neutralizing these frequency
variations, the detected signals derived from the varying group of
resonances are shifted to the outputs of said channels in an order
that the signal derived from the lowest (pitch) frequency in the
group is shifted to the number ONE output of the numerically
arranged bank of channels, while the signals derived from the other
resonances are shifted to the same mutually related numerical ratio
separations along the channel outputs as differ in mutually related
frequency ratios of the original resonances. Thus if the lowest
frequency in a group is 600 cycles per second, it is necessary to
start from the signal-regrouping combination that will effect
shifting the detected signal derived from the sub-band frequency of
600 cycles per second to the output of number ONE channel. In order
to avoid any time waste in testing of all the filter outputs from
arriving frequencies below 600 cycles per second, a fast
signal-hunting distributor is used which scans all of the detected
signals derived from the sub-bands starting from the lowest in the
total number of sub-bands, and allow operation of the respective
signal-regrouping combination only when a detected signal at the
output of the pass-band filter responding to 600 cycles per second
(exemplary) is present. With the modern available integrated
circuitry, the hunting system can be made as fast as desired, so
that the time consumed for hunting can be made negligible. Then
again, since the information bearing signals are within a region
above the pitch frequency (as explained in my reference patent), it
is only necessary to perform two or three more (or other if so
desired) signal regrouping operations after the detected signal
derived from the pitch frequency has been located by the
distributor, for selecting a specific group of signals that
represent the spoken phonetic sound. This is accomplished by a
signal counter, which resets the distributory (hunting) system into
normal operating state after four counts of signal-regrouping
operations have been completed. Thus the object of the present
invention is to provide improved and practical simplicity of
operation over the system disclosed in my reference patent. Another
object of the invention is to provide an arrangement for alternate
operation during complete succeeding pitch periods, so that greater
time during the pitch periods can be devoted to signal-regrouping
operations. Still another object is the specific sub-division of
the sound spectrum by pass-band filters, for obtaining highly
precise frequency conversion of the sound wave.
In my previous disclosures of making automata for phonetic sound
recognition, I have described systems both for direct analysis of
the sound wave, such as described in my related U.S. Pat. No.
3,322,898 May 30, 1967, and modifying the frequency variations of
the sound wave into a normalized form prior to analysis, so that
the analyzing process can be standardized; in both cases the
resulting performance being the same. In the system disclosed
herein, however, normalization of frequency variations is preferred
for practicability purposes, and since it is claimed herein to be
free of environmental changes, it is also claimed to perform the
ultimate in man made automata for phonetic sound recognition. Thus
in order to draw a distinction between the highly accurate
performance of the system disclosed herein and the systems given in
prior teachings, it is first necessary to show how closely the
analytical process of the presently given system simulates the
analytical process of the human brain -- even though their physical
performances may differ widely.
THEORETICAL BRIEF RELATING TO THE INTERPRETIVE FUNCTION OF THE
BRAIN
I have given a detailed description of the analytical function of
the human brain in my first reference patent. Thus it is not
necessary to repeat herein the complete text of my former
description, and the following brief will be sufficient for the
computer engineer to visualize a series of processing steps that
will closely simulate the biological aspects involved in the
interpretation of phonetic sound waves.
In order for the brain to be able to interpret sound informations
intelligibly, it must be informed as to when to start analysis and
when to end analysis of the arriving information. In sound
information, the major peaks of the sound waves are the indicating
points between which the sound waves are analyzed and
re-constructed into the original sound information. For example, in
phonetic sounds, there are high-peaked major peaks, and sub-peaked
major peaks. The waves contained between the high-peaked major
peaks represent both the phonetic and quality informations of the
sound. Whereas, the waves contained within the sub-major peaks of
the sound represent only the phonetic information. These major
peaks are initiated by puffs of air from the glottis into the mouth
cavities. For example, as each puff of air enters the specifically
formed mouth cavities, it forms an initial high peaked wave,
representing the high-peaked major peaks. According to the specific
formation of the mouth cavities, the initial air force in the mouth
sets up several resonances which have definite frequency and
amplitude ratios one with respect to another (these resonances do
not have fixed frequency distribution in the sound spectrum, as
believed conventionally, but fixed ratio relationships,
representing the phonetic sound). As each puff of air is forced
into the mouth cavities, the vocal folds (vocal cords) also start
vibrating at a frequency as determined by their physical
structures, and the physical tension imposed upon them. Thus
besides the high-peaked puffs of air from the glottis, there are
also subpeaked puffs of air from the vocal cords that enter the
mouth cavities.
The time period between each high-peaked major peak is the pitch
period of the sound, and the time period between each subpeaked
major peak is the fundamental time period of the sound. As
indicated above, these peaks are absolutely necessary for the brain
to be able to interpret the arriving phonetic sound. This is the
reason why a person who has lost his voice box by surgery, is able
to produce intelligible phonetic sounds in speech, by forming his
mouth slowly to the phonetic sounds that he normally would for the
speech, and pressing a mechanical vibrator against his throat; the
vibratory lobes representing the major peaks. Thus the major peaks
are not part of the phonetic sound, but their presence is
absolutely necessary for the brain to know when to start and when
to conclude the analysis of the information presented. For only
phonetic sound recognition then, those resonances that occur above
the fundamental are selected and analyzed by frequency and
amplitude ratio measurements; an illustrative description of which
would be too long to include herein, but reference may be made to
my first reference patent. For practical simulation of the brain's
interpretive function, the frequency variations that normally occur
in voices of different speakers are first normalized prior to
analytical processing, a detailed description of which will now be
given in conjunction with the accompanying drawings, wherein:
FIG. 1 is partly schematic and partly block diagram of the complete
system of speech typewriter according to the invention.
FIG. 2 is a waveform illustrating the alternate READ and WRITE
operating periods of the system in FIG. 1.
FIG. 3 is an exemplary signal-amplitude-ratio measuring arrangement
according to the invention.
FIG. 4 shows the center frequency sub-divisions of the pass-band
filters, as used in the present invention.
FIG. 5 is a numerical chart showing how the detected signal outputs
of the pass-band filters are switched linearly to the outputs of
the numerically arranged channels, in accordance with the
invention.
FIG. 6 is a switching arrangement using analog channel switches,
showing the simple parallel connected gate electrodes of the
switches in accordance with the specific sub-band divisions of FIG.
4.
And FIG. 7 shows how these analog switches can be oriented on a
planar surface for simple interconnections of terminals, in
accordance with the invention.
In order to obtain high accuracy of signal regrouping without
causing any cross switching of the input signal to the channel
outputs, the center frequencies of the sub-dividing band-pass
filters are arranged as shown in FIG. 4, wherein the frequency
sub-divisions are similar to the standard musical scale. Such a
numerical arrangement simplfies the actual channel switching
arrangement, because it requires linearly sequenced numerical
transfer without cross coupling of any of the pass-band outputs,
such as would be by the arrangement of pass-band center frequencies
of the filters shown in FIG. 11, and the chart of switching
combinations in FIG. 12 of my first reference patent, as mentioned
in the foregoing. This is shown in greater clarity by the chart in
FIG. 6, wherein the top row of the numerals represent the channels,
and the row of numerals below represent the sequence of the
numerals left hand of the sub-band frequencies in FIG. 5. For
example, in the first row (FIG. 6), all of the detected outputs of
pass-band filters (starting from filter number-1) are applied to
the inputs of the channels starting from channel number- 1. In the
second row, the detected filter outputs starting from the number- 2
filter are applied to all of the channels starting from the channel
number- 1, and so on. As stated in the foregoing, such simplicity
of switching sequence becomes inherently accurate, as long as the
harmonic sequence at like intervals is considered in sub-dividing
the sound spectrum, because the number of sub-band divisions and
center frequencies of the pass-band filters may be arranged other
than the frequencies shown in FIG. 4 without affecting the required
accuracy. In furtherance, the numerical sequence assigned to the
pass-band filters may be reversed, or coded, according to any mode
of analytical switching of the sub-band signals that may be
desired.
THE SWITCHING ARRANGEMENT OF FIG. 6
As stated in the foregoing, the specific sub-band division of the
sound spectrum simplifies the switching arrangement. Due to the
plurality of parallel connections that are necessary in physical
assembly, it is preferably to use high impedance transistors, such
as the MOS FET transistor, which are particularly simple for
integrated assembly on single wafers, but not necessarily, if
practice indicates otherwise. The circuitry of the channel
switching may be arranged as in FIG. 6, wherein, the channel-1 is
represented by the transistors Q21, Q23, Q26 and Q28; channel- 2 is
represented by the transistors Q20, Q24, Q27; and the channel- 3 is
represented by the transistors Q21, Q25; and the channel outputs
are represented by the load resistors R1 through R3*, respectively.
Such an arrangement provides plurality of inputs for each channel,
for example, the input of channel-1 may be switched to either the
output of DR1, DR2, DR3 arriving at the inputs of block 4 in FIG.
1; this also being applied to the nth channel. The switching
arrangement shown in FIG. 6 is similar to the switching arrangement
shown in FIG. 10 of my first reference patent, but it is simpler in
physical assembly.
The switching sequence is obtained by parallel connections of the
gate electrodes in the numerical sequence as shown in FIG. 5. For
example, in the first row of parallel connections, the gate
electrode of Q19 (channel-1) is connected to the gate of Q20
(channel-2), and to the gate electrode of Q21 (channel-3). In the
second row of parallel connections, the gate electrode of Q23
(channel-1) is connected to the gate electrode of Q24 (channel-2),
and so on. When these parallel connected terminals of the gate
electrodes are normally biased in backward direction, all of the
switches remain in OFF states. Anyone of these parallel connected
terminals receiving a forward bias, however, the outputs of a set
of the pass-band filters are switched to the channel outputs, as in
the arrangements of FIG. 5. The parallel connected gate terminals
may either be actuated sequentially or randomly, for the desired
effect of signal regrouping.
SWITCHING APPARATUS OF FIG. 7
In reference to the chart shown in FIG. 5, it is apparent that the
use of a large numer of switching transistors is required for the
switching performance. While multiple element channel switches in
small packages of integrated circuits are available, it still will
require large surface of printed circuit board with a large number
of terminals to be soldered. Due to the simplicity of
interconnecting design, however, a large number of these
transistors, or any type of controllable semiconductors, may be
integrated on a single wafer, such as shown in FIG. 7, wherein,
each square represents the semiconductor as mentioned, three
squares of which are shown in shaded lines in order to render them
distinguishable in the maze of electrical connections. The
connecting electrodes in each square are represented by the black
dots to which are terminated the required parallel connections, as
illustrated by the horizontal, vertical, and 45.degree. parallel
lines; the first representing the source electrode terminals, the
second representing the drain electrode terminals, and the third
representing the gate electrode terminals.
Having described the details of frequency normalization, or
signal-regrouping, as utilized herein, the complete arrangement of
a phonetic sound recognizing apparatus will now be described by way
of the part shematic and part block diagram in FIG. 1, as in the
following:
PHONETIC SOUND RECOGNIZING APPARATUS OF FIG. 1
As had been mentioned in the foregoing, alternate operation during
succeeding pitch periods in the sound wave are used in the
arrangement of FIG. 1, so as to allow as much signal-regrouping and
analyzing time as possible. Thus by using first and second branches
in the system for alternate operation, the various components in
the sound wave are stored (WRITE) alternately in the first and
second branches during successive pitch periods, and these stored
signals are analyzed and discharged alternately during the (READ)
periods. Thus the system in FIG. 1 consists of: (1) a plurality of
pass-band filters for dividing the sound wave into sub-bands; (2)
two separate storage devices at the output of each filter for
storing signals alternately during succeeding pitch periods, by an
alternate switching arrangement at pitch periods; (3) a frequency
normalyzing (signal-regrouping) arrangement as described in the
foregoing; (4) a plurality of coupling means from the alternate
storage signals to the frequency normalyzing arrangement for
regrouping the various stored signals in a group; (5) a signal
distributor for operating the signal regrouping arrangement
sequentially during the (READ) period until the proper signal
regrouping is established; and (6) means for matching mutually
related amplitude ratios of the regrouped signals during the (READ)
period with that of prearranged amplitude ratio measuring means for
final decision of the information sought; following discharge of
the stored signals that had been analyzed.
STORAGE SYSTEM OF THE SUB-BAND SIGNALS IN ALTERNATE PITCH
PERIODS
In FIG. 1, there are shown only three pass-band filters in blocks
1, 2 and 3, The operation of all these sub-band filters in
conjunction with their associated circuitry is the same, and
therefore, the operation of only the filter in block 1 and its
associated circuitry will be described, as a typical example. Thus
the output of sub-band frequency f.sub.1 in block 1 is divided into
two separate branches across the center tapped secondaries L1 and
L2 of transformer T1. The output of L1 is full wave rectified by
the diodes D1, D2, and stored across capacitor C1 in series with
the charging transistor Q1, while the output of L2 is full wave
rectified by diodes D3, D4, and stored across C2 in series with Q4.
The normally idle transistor Q2 is connected in parallel with C1,
so that the stored potential can be discharged after READ period
has ended, and the normally idle Q4 is connected in parallel with
C2 for discharging the stored potential during an alternate time
period with respect to the first discharge. The stored signals
across C1 and C2 are switched alternately during alternate READ
periods to the common output of transistors Q13 and Q14, which are
excited into on-and-off states alternately during the succeeding
(READ) periods. This common output is connected directly to
terminal DR1 of the signal regrouping block 4. The output of
pass-band filter in block 1 is also applied to an amplitude
threshold sensing device in block 5 (which can be obtained
commercially in the form of integrated circuitry), and applied to
the R-S (set-reset) flip-flop in block 6 for operation. Thus when a
signal appears at the output of block 1 just above a threshold
level, the R-S flip-flop in block 6 operates in set state,
indicating that a signal has appeared at the output of the filter
in block 1. This set-operating signal of block 6 is applied in "1"
level to one of the inputs of AND gate in block 7, but it does not
operate until its second input is also driven to 1 level. When the
output of AND gate 7 becomes active, it operates the one-shot (OS)
in block 8, which in turn produces an output pulse of predetermined
time length and applies to one of the multi-inputs of gate in block
9, and also to the lower left hand terminal 1 of the signal
regrouping block 4. In similar fashion of operating conditions of
the filter blocks 2 and 3, the common output of the transistors
Q15, Q16 is applied to the terminals DR2 and DR3 of the block 4,
and the output pulses of the one-shots in blocks 13 and 17 are
applied to the inputs of gate in block 9, and also to the lower
center and right hand terminals 2 and 3 of block 4, respectively.
The DR1, DR2 and DR3 terminals of block 4 in FIG. 1 represent the
drain electrode terminals of the arrangement in FIG. 6, and the
numerals of the lower end terminals of this block arrangement
represent the same numerical terminals of the gate electrodes in
the same arrangement. Thus referring to the electrical connections
just described, and starting from the lowest frequency in the
sub-band division, if an arriving signal appears at the output of
filter block 1, the output pulse of one-shot in block 8 switches
the transistors Q19-Q22 (FIG. 6) in ON states for regrouping the
stored signals at the input of block 4. In the event that the
output of filter in block 1 is zero, and a signal appears at the
output filter in block 2, then the output pulse on one-shot in
block 13 switches the transistors Q23-Q25 in ON states for
regrouping the stored signals at the inputs of block 4. Similarly,
if the outputs of blocks 1 and 2 are zero, and a signal appears at
the output of block 3, then the transistors Q26-Q27 are switched in
ON states for regrouping the stored signals. The object at this
point, accordingly, is to provide a hunting arrangement in the form
of a distributor, which starts distributing 1 level pulses to the
second inputs of AND gates in blocks 6, 11 and 15, starting from
block 6, or in any sequence that may be desired, so that whichever
AND gate has one of its inputs driven into 1 level by the output of
its associated pass-band filter, its following one-shot operates by
the distributor pulse for the prearranged signal regrouping. During
any of these signal regrouping operation, the mutually related
amplitude ratios of these regrouped signals are matched with
prearranged ratio measuring groups to make sure that the proper
signal regrouping has been established for recognition of the sound
information. If not, however, the signal regrouping continues until
the correct regrouping produces an output signal representing the
information sought. In order to save time in these regrouping
operations, the distributor pulses are made much shorter than the
output pulses of the one-shots, and the distribution frequency is
made practically high enough to skip the inoperable AND gates
(blocks 6, 11, 15) in negligible length of time. During the
operation of any one-shot, of course, the distributor is inhibited
for the required analytical performance.
Having described the general purpose of the arrangement in FIG. 1,
details will now be given, as in the following:
ALTERNATE SWITCHING SIGNALS
It had been described above that the capacitors C1 and C2 are
charged and discharged alternately during succeeding pitch periods
in series with Q1 and Q3 respectively, and the charged capacitors
are switched to the common output of Q13 and Q14 alternately for
analysis. The sequence of this operation is illustrated in FIG. 2,
wherein during first part of the READ period the Q1 is in OFF state
for transmitting the charge of C1 to the block 4. After the READ
period has ended, the Q13 becomes in OFF state, and the Q2 becomes
in ON (shaded) state for discharging the capacitor C2. During this
READ period, the Q3 becomes in ON state for charging the capacitor
C2, while the Q14 becomes in OFF state for preventing any transfer
of this charging action to the common output of Q13 and Q14. The
READ period of the charge of C2 is shown by the ON period of Q14
being in alternate periods of the Q13. Similarly, the discharging
of the capacitor C2 is shown by the ON (shaded) periods of the Q4
in alternate periods with respect to the Q2. With this sequence of
alternate operations of the charging and discharging transistors to
the capacitors C1 through C6, it is now necessary to provide the
switching signals to these transistors, as in the following:
In FIG. 1 the voice sound wave in block 18 is applied to the pitch
selector in block 19 (which may have a one-shot at its output for
producing sharp edged pulses) for deriving pulse signals at pitch
frequencies (major peaks). These pulse signals are applied to the
clock flip-flop in block 20 for producing alternate output signals
in steady state steps, which represent the waves of Q1 or Q3 in
FIG. 2. The steady state output signals of block 20 are applied
directly and alternately to the first inputs of AND gates in blocks
21 and 22, and further also applied alternately to the inputs of
set-reset flip-flops in blocks 23 through 26 by way of direct and
the differentiating coupling capacitors C7 through C10,
respectively. The purpose of using the coupling capacitors C7
through C10 is to avoid steady state excitation at the inputs of
these flip-flops. The second inputs of AND gates 21 and 22 are
connected in parallel, and normally biased in backward direction so
that signals in forward direction at their inputs alone will not
render them operative. Thus assuming during one pitch period of the
incoming sound wave that the upper terminal of clock flip-flop 20
is in positive polarity, it prepares the AND gate 21 ready for
operation when a forward bias pulse upon its second input arrives.
The lower terminal of block 20 applies a negative pulse to the
input of set-reset flip-flop 26 to operate it in set-position for
producing at its output the wave of Q13 in FIG. 2. After a certain
signal-distribution time period (the operation of which will be
described further by the distributor arrangement), a pulse signal
in forward direction arrives at the parallel connected second
inputs of AND gates 21 and 22, causing the gate 21 to operate and
apply a pulse signal in forward direction to the inputs of
set-reset flip-flops 23 and 25. The flip-flop 23 is driven into
reset-position (so as to terminate its READ period), and
simultaneously drives the flip-flop 25 into set-position, which
produces at its output the shaded portion (storage-discharge) of
the wave Q2 in FIG. 2. When the clock flip-flop in block 20
reverses its state of operation by the arriving pitch pulse, the
input of set-reset flip-flop 25 receives a pulse in forward
direction from the upper terminal of flip-flop 20, and it shifts
into reset-operation for terminating the shaded portion of the wave
Q2 in FIG. 2. Simultaneously, the set-reset flip-flop in block 24
operates in set-position for repeating the previous operation in
alternate cycles.
The terminals of flip-flops 23-26 are shown connected to the
transistor control gates of Q1 through Q18, in series with the
independent amplifiers in block 27. By using the present day
component parts and integrated devices, the diagram in FIG. 1
indicates that the transistors Q1 through Q18 are of the MOS FET
type which require about -10 volts at the control gates for
operation. Whereas, if the flip-flops used are of the type using
about 5 volts for operation, there are also available integrated
circuit amplifiers for the purpose of interfacing with high and low
operating devices, such as used in the drawing of FIG. 1, although
different types of available devices may also be used, if so
desired. The amplitude equalizer in block 28 is also shown, as used
herein, for normalizing the amplitude variations of the sound wave
prior to analysis, and it may be of any available type, or as
described in my previous patent issues. Thus having described the
alternate switching arrangement, the signal distributing (hunting)
system may now be described, as in the following:
DISTRIBUTOR ARRANGEMENT OF FIG. 1.
As had been described in the foregoing, signal regrouping occurs
when the first and second inputs of anyone of the AND gates 7, 12
and 16 are simultaneously driven into 1 level voltages. Thus the
purpose of the distributor is to apply distributory 1 level signals
to the second inputs of these gates (as a hunting process), so that
the AND gate which has received 1 level signal at its first input
will operate, and thereby causing its associated one-shot to follow
operation for the required signal-regrouping action in block 4. The
distributor is represented by the blocks 28 and 29, which for
simplicity of design may be two of the available
four-line-to-16-line-decoders connected in series for 32
distributory output pulses. The outputs of a binary counter are
connected to the A, B, C, D inputs of the blocks 28, 29, so that
the sequential pulses applied to the input terminal 5 of the binary
counter 30 are converted into sequentially distributed pulses along
the total of 32 independent outputs of the blocks 28 and 29. These
outputs (phase inverted but not shown) are applied individually to
the second inputs of the AND gates, only three of which are shown
in the drawing. The use of 32 distributory outputs is shown for
versatility of the arrangement in FIG. 1 for various other uses,
and the number of these outputs may be more or less according to
the type of complex wave analysis used for.
The input terminal 5 of binary counter in block 30 is driven by the
clock generator in block 31 (for example, at a frequency of 300
KHz) in series with the gate in block 32 and inverter block 33. The
three inputs of the gate are normally at 1 level, so that the 1
level clock pulses are admitted to the input 5 of the counter 30
for operating the distributor. Thus assuming in one example that
the first input of gate 7 has received a signal at 1 level from the
R-S flip-flop in block 6, the first distributor pulse (at 1 level
after being phase inverted, not shown) applied to the second input
of AND gate 7 causes operation of the one-shot 8, which in turn
applies a pulse of 0.2 millisecond at "0" level to the input of
gate 32 (in series with the multi-input gate 9 and the 3-input gate
34) to render it inoperative, and thereby stop the clock pulses
passing to the input of counter 30 for the first combination of
signal regrouping operation in the block 4. In a second example,
assuming that the first input of gate 7 is at 0 level, and the
first input of the gate 12 is at 1 level, the one-shot 8 does not
operate, but the second distributory pulse causes operation of the
one-shot in block 13. The output pulse of 0.2 millisecond long
stops transmission of the clock pulses to the input of counter 30,
and now the second combination of signal-regrouping is processed in
the block 4. Such a performance illustrates the time saving in the
high frequency hunting (distributory) system, wherein, the time
delay of 0.2 millisecond that would normally be required for the
operation of one-shot to the next, is skipped during only 8
microseconds during hunting periods. As described in the foregoing,
however, the operation of one of the one-shots 8, 13, 17 . . . n,
may not be sufficient for the correct signal regrouping, and
several more of the number of signal-regrouping operations may be
necessary before the correct signal regrouping has been
established. Thus a counter is also included herein, so that it can
reset the distributor after a predetermined number of counts,
because more than a predetermined number of signal-regrouping
counts is indicative of lack of information present. Thus the
specific details of the counter in connection with the distributor
is explained, as in the following:
In detailing the operation of the distributor, and starting from a
pitch pulse, assume that the pitch pulse operates the one-shot in
block 35 which triggers and applies a resetting pulse to the clear
terminal 14 of binary counter 36, and to the R-S flip-flop 37 which
applies 1 level signal to one of the inputs of gate 32. The normal
output of gate 34 to one of the inputs of the gate 32 is also at 1
level, so that the clock pulses pass on to the counter 30 for
operation. The counter 30 triggers during the input low-to-high
transition period, which allows time for resetting itself first.
The R-S 40 also operates and resets the distributor in block 28
into normal operating state, while at the same time inhibiting the
block 29. The counter 30 now starts operation and the block 28
starts distributing 1 level pulses (after inversion, not shown) to
the first inputs of the AND gates 7, 12, 16 . . . n. At the end of
16 distributory counts, the counter 30 resets itself by a pulse
from its carry output 12 to th clear input 14, via gate and
inverter in block 39, and triggers the R-S 40 which in turn
inhibits the block 28 and activates the block 29, for a total of 32
distributory outputs.
After a predetermined number of counts by the binary counter in
block 36 (or any other type of counter), it triggers the R-S
flip-flop in block 37, which inhibits the AND gate 32, and applies
1 level signal to the inputs of AND gates in blocks 21 and 22 (in
series with auxiliary gate 41) to start discharge of one of the
pairs of storage capacitors (C1 or C2), as a new cycle of operation
by the following pitch pulse. The counter 36 also sends a pulse
signal to the signal-decoder block 42, (signal-amplitude ratio
measuring circuits) which when activated by an incoming sound
information (speech) a typewriter key is actuated in block 43, as a
final translation from sound to visible indicia.
In the case that the distributor in block 29 operates to the nth
output terminal without causing operation of the counter in block
36 for reset cycling, the signal from the nth terminal (y) is
connected to the input of gate 41 to effect discharge of the
capacitors C1-C6, so that recycling of the arrangement can start on
the following pitch pulse.
SIGNAL-AMPLITUDE-RATIO MEASURING CIRCUIT OF FIG. 3
As described in the foregoing, after selection of a group of stored
signals from the outputs of the signal-regrouping arrangement, the
amplitude ratios of these signals one with respect to another are
further matched with predetermined ratios for final decision if the
selected group of signals represent a specific sound information.
The amplitude ratio measuring arrangement shown herein in FIG. 3 is
similar to the arrangement shown in FIG. 14 of my disclosure in my
U.S. Pat. No. 3,622,706 Nov. 23, 1971, with modified component
parts for the sake of greater stability that be obtained for
practical purposes. Thus assuning that the amplitude ratio between
the signals A and B (FIG. 3) has to be matched with that of a
predetermined ratio, the A signal is applied to the transistor Q29
and the B signal is applied to the transistor Q30. As had been
described by way of the one-shots 8, 13, 17 . . . n, the selected
group of signals are in ON states within only 0.2 milliseconds,
which cause proportional currents through the primaries of the
transformers T4 and T5. One of the terminals of the secondaries of
T4 and T5 are connected to ground in series with the bias source
B1, and the other terminals are labeled as OUT representing the
output terminals. The secondaries of T4 and T5 are also shunted by
diodes D13 ad D14, respectively, in series with the bias source B2.
These diodes are used to prevent oscillation in the secondaries
after initiating a pulse current (but may be omitted with the B1
and B2 if not found necessary), and the voltage levels of bias
sources B1 and B2 are adjusted equal to the conducting threshold
gaps of the diodes D13 and D14, so that the diodes will start
conducting from close to zero voltage level across the secondaries
of T4 T5.
The output of the secondary of T4 is connected to ground in series
with one of the signal-mixing diodes D15 through D17, and resistor
R4, and the output of the secondary of T5 is connected to ground in
series with one of the signal-mixing diodes D18 to D20, and
resistor R5. For each specific sound information the voltage gains
across R4 and R5 are preadjusted, and applied to the gate
electrodes of transistors Q31 and Q32, respectively. The resultant
effect is that, the oppositely polarized voltages across
secondaries of T6 and T7 will either nullify to zero voltage for a
specific gain ratio adjustments across R4 and R5, or above zero
voltage (positive or negative) when the incoming information is
other than the fixed gain adjustments. Finally, the outputs of
series connected secondaries of T4 and T5 are applied to the gate
electrode of amplifier transistor Q33, and the secondary of T8 is
full wave rectified by the diodes D21, D22 for application in 0
level to one of the inputs of the AND gate in block 44. The other
input of this gate is normally biasd to 0 level, so that the output
of gate 44 will not operate the one-shot in block 45, until both
inputs of gate 44 are at 1 levels. Thus when the output pulse from
the gate 32 in FIG. 1 is applied to the one-shot in block 46 of
FIG. 3, it operates with a delay pulse and further operates the
one-shot in block 47 through the differentiating coupling capacitor
C11. The output pulse of one-shot 47 is finally applied to one of
the inputs of gate in block 44 in 1 level. If at this time the
other input has not received 0 level voltage from the secondary of
T8, and has remained at 1 level, the gate 44 operates the one-shot,
in block 45 for the required typing of a letter symbol. On the
other hand, if the gate 44 has received 0 level signal from T8, it
remains inoperative by the arriving pulse from one-shot in block
47. The diode 21 is used to prevent excessive voltage applied to
the gate 44, but may be dispensed with without affecting the
operation of the system. Similarly, amplitude-ratio measuring
arrangements other than shown herein may be used as long as it
serves the purpose of the present invention, for example, voltage
comparing circuits such as commercially available in the form of
integrated devices may also be used. The circuit shown in FIG. 3,
however, is simple, and because of the transformers used for
obtaining null signal, any number of transformer secondaries may be
inserted in series with the secondaries of transformers T6 and
T7.
While specific embodiments of the invention have been selected to
describe the invention, it is obvious to the skilled in the art
that may be considered as exemplary, and that the invention is not
limited in its utility. Accordingly, it is also obvious that
various modifications, adaptations, and substitutions of parts may
be made without departing from the true spirit and scope of the
invention.
* * * * *