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BACKGROUND OF THE INVENTION
The present invention relates to a voice modifier, and more particularly
relates to a voice modifier which can transpose or distort one voice into
another voice by receiving the input audio signals with normal frequency
and transmitting the output audio signals with different or unusual
frequencies.
Heretofore, the concepts involving speech sampling and companding, and the
variation of the frequencies in clock signals have been developed in many
parts of the world. For example, U.S. Pat. No. 4,682,362 to DeFreitas et
al., entitled "Generating Narrowly-separated Variable-frequency Clock
Signals", discloses circuitry for generating clock signals of slightly
different frequencies in which an analog to digital converter, a memory
and a digital to analog converter are provided. A primary objective of the
prior art is to produce a flange effect between the analog output signal
and input signal. This is achieved by the employment of at least one of
the clocks having a variable frequency provided by a voltage-to-frequency
converter driven by the output of an integrator, which is supplied with
the sum of a frequency-difference-command signal (supplied by a
supervising microprocessor) and a difference signal representative of the
actual difference in frequency between the two clocks. Data is written
into memory at the address specified by encoding pointer, and data is read
from the memory at the address specified by decoding pointer. It should be
noted that the speed of the encoding pointer is set by fixed encoding
clock and is constant; changes in delay are brought about by slightly
varying the decoding clock relative to the encoding clock, under control
of a microprocessor. Comprehensively, the decoding pointer has to trail
behind the encoding pointer by the time period of the delay; moreover, the
longer the of delay is, the more memory capacity is needed to store the
signals.
Unlike the prior art, which discloses circuitry to control the clocks in a
system for providing variable delay of an audio signal, the encoding
pointer and decoding pointer used in the present invention are independent
and different to produce a transposing or distorting effect on audio
signal. The features of this invention can be easily realized by
explaining the operation of a phonograph.
As is well known, if a phonograph record is played at 331/3 RPM, it will
create a much different effect than if it is played at 45 RPM. So
obviously, one means of modifying the quality of a voice would be to
transmit the audio signals at a frequency higher or lower than the
receiving frequency, for example, twice or half the receiving frequency.
For the purpose of further understanding the teaching of this invention,
the basic concept and basic block diagram thereof are shown hereinafter
with brief description.
FIG. 1A is a graph illustrating the relation between voltage and time of an
input signal waveform, and FIG. 1B is a similar graph with a transmitting
frequency equal to twice the transmitting frequency of FIG. 1A. When at
time T, the position of voltage of FIG. 1B with respect to time is
equivalent to the position of voltage of FIG. 1A corresponding to time 2T.
More specifically, FIG. 2 schematically illustrates a signal process for a
voice modifier comprising an analog to digital (A/D) converter 62 which
converts an analog signal 61 into a digital signal, a memory unit 63 which
stores the digital signal, a digital to analog (D/A) converter 64 which
converts the stored digital signal into an analog signal, and an amplifier
65. The voice modifier has a transmitting frequency f2 higher/lower than
the receiving frequency f1 so that the output signal is different from the
input signal.
The concepts of over-write and over-read are involved in this invention to
solve the demand of an infinite memory when the encoding pointer is far
ahead the decoding pointer (i.e., receiving frequency>>transmitting
frequency) and the stack of digital signals being stored is increasing
with time, since the decoding pointer is independent from the encoding
pointer. Moreover, a continuous variable-slope delta modulation (CVSD),
adaptive delta modulation (ADM), or delta modulation (DM) system is
employed in this invention for designing D/A and A/D converters to smooth
the discontinuous portions resulting from over-write or over-read.
SUMMARY OF THE INVENTION
A primary objective of the present invention is to provide a voice modifier
which can transpose or distort one voice into another voice.
Another objective of the present invention is to provide a voice modifier
which can be applied to telephones, microphones, loudspeakers and so on.
Still another objective of the present invention is to provide a voice
modifier which, when mounted on a telephone, prevents the person receiving
the call from recognizing the caller's voice.
Further objectives and advantages of the present invention will become
apparent as the following description proceeds, and the features of
novelty which characterize the invention will be pointed out with
particularity in the claims annexed to and forming a part of this
invention.
BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1A is a graph illustrating the relation between voltage and time for
an audio input signal waveform;
FIG. 1B is another graph illustrating the relation between voltage and time
for an audio input signal waveform similar to that shown in FIG. 1A, but
with a higher transmitting frequency;
FIG. 2 is a basic block diagram for processing an audio signal;
FIG. 3 is a block diagram illustrating a signal process for voice
modification;
FIGS. 4A to 4D are diagrammatic representations showing the relation
between the encoding pointer and decoding pointer, in which the encoding
pointer is ordinarily ahead of the decoding pointer;
FIGS. 5A and 5B respectively represent the written waveform and read
waveform with reference to FIGS. 4A to 4D;
FIGS. 6A to 6D are another group of diagrammatic representations similar to
FIGS. 4A to 4D, but wherein the decoding pointer is ahead of the encoding
pointer;
FIGS. 7A and 7B are waveforms similar to FIGS. 5A and 5B, but with
reference to FIGS. 6A to 6D;
FIG. 8 is a graph illustrating a series of input audio signals with respect
to voltage and time, wherein the input audio signal is divided into
several sections with waveform 1 at interval t1, waveform 2 at interval
t2, waveform 3 at interval t3, waveform 4 at interval t4, waveform 5 at
interval t5, and so on for sake of convenience;
FIG. 9 is another graph similar to FIG. 8, but illustrating that the
waveforms 1-4 are repeated at intervals 1-4, respectively, with
transmitting frequency of FIG. 9 equal to twice that of FIG. 8;
FIG. 10 is still another graph similar to FIG. 8, but illustrating that
waveforms 1-4 are cut in half and amplified twice at intervals 1-4,
respectively, with transmitting frequency of FIG. 10 half to that of FIG.
8;
FIG. 11 is another graph illustrating that waveform 1 is repeated
separately at intervals 1, 2, 3 and 4, and illustrating that waveform 5 is
repeated separately at intervals 5, 6, 7 and 8; and
FIG. 12 is a waveform for a second encoding clock 2 and a second decoding
clock 2, wherein a pulse of the second decoding clock is not spaced from
its preceding pulse and from its subsequent pulse by the same time
interval.
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT
With reference to the drawings and particularly to FIG. 3 thereof, a block
diagram illustrating one preferred embodiment of an electrical device of a
voice modifier in accordance with the present invention can be seen. The
process used in FIG. 3 is similar to that shown in FIG. 2 and further
similar to FIG. 1 of U.S. Pat. No. 4,682,362, as heretofore stated. For
example, analog to digital converter 13 converts analog input signal to a
digital signal, which is written into a memory unit 14 (a static random
access memory, a dynamic random access memory or a register) at the
address specified by an encoding pointer EPOINTER. After control
processes, which will be detailed hereinafter, the analog output signal is
generated by a digital to analog converter 15, using digital output
signal, which is read from memory unit 14 at the address specified by a
decoding pointer DPOINTER. The analog input signal is preprocessed by a
first amplifier 12 via a low-pass filter from a microphone 11, and the
analog output signal from the D/A converter 14 is further processed by a
second amplifier 16 and then transmitted by a loudspeaker 17 (or the
like).
A multiplexer 20 alternately connects DPOINTER and EPOINTER to the address
input of the memory unit 14. The encoder pointer EPOINTER is created by a
write address counter 21 which is incremented at an appropriate period by
encoding clock ECLK, supplied by a programmable logic array 25. Similarly,
the decoder pointer DPOINTER is created by a read address counter 22 which
is incremented at another appropriate period by decoding clock DCLK,
supplied by the programmable logic array 25. Furthermore, the encoding
clock ECLK and the decoding clock DCLK are respectively sent to the A/D
converter 13 and the D/A converter 15 to actuate the operation of
converters 13 and 15.
A clock generator 3 which operates the control processes comprises an
oscillator 31, a timing generator 32 and the programmable logic array 25
to generate two different clocks, namely, the encoding clock and decoding
clock. The programmable logic array 25 is further controlled by a switch
array or push button 40 for adjusting the decoding clock with respect to
the encoding clock. The switch array 40 utilizes an oscillator, a variable
resistor means or a divider to generate the decoding clock different from
the encoding clock. Since switch array technology and construction are
well-known in the art, further discussion and description are not
considered necessary at this time. The clock generator 3 additionally
comprises a timer 33 for instructing the digital to analog converter 15 to
transmit a signal indicating a silence via a control logic block 34 when
no signal with sufficient intensity is read by A/D converter 13 for a long
period of time.
The concepts of over-write and over-read are employed in this application
to solve the previous problem demand of an infinite memory requirement,
since the transmitting (converting) frequency of the decoding clock DCLK
is possibly ahead of the receiving (sampling) frequency of the encoding
clock ECLK. Over-write (or over-read) means that when the address pointer,
which designates the address position of data to be written (or read) in
the write address counter 21 (or the read address counter 22), points to
the highest byte (i.e., FFFF) of the memory, the address pointer will
automatically point back to the lowest byte (i.e., 0000) of the memory
when the address is incremented by one. This makes a cycle for the memory
block and results from the transmitting frequency being independent from
the receiving frequency; however, the transmitting and receiving
frequencies have to be defined within an acceptable range.
For convenience of explanation, an unit cycle with the EPOINTER and
DPOINTER is used to simulate the data written into and read from the
memory. Operation of the preferred embodiment can be understood with a
first situation shown in FIGS. 4A to 4D in view of 5A and 5B. Data is
written into the memory at the address specified by encoding pointer
EPOINTER, and data is read therefrom at the address specified by decoding
pointer DPOINTER. In the first situation, when receiving (sampling)
frequency RF1>transmitting (converting) frequency TF1, for example, RF1=2
TF1, FIGS. 4A to 4D diagrammatically represent the relation therebetween.
Obviously, EPOINTER and DPOINTER would meet again after EPOINTER traveling
two rounds of the unit cycle and DPOINTER traveling one round of the unit
cycle in which "a round" shown in FIGS. 4A to 4D can be considered as the
period from "t1 to t2" or "t3 to t4" shown in FIGS. 5A and 5B. As can be
seen in FIGS. 5A and 5B, the waveforms in FIG. 5A are cut in half and then
amplified twice in the corresponding periods of FIG. 5B due to the
difference between the receiving and transmitting speeds.
Similar to the first situation, a second situation, when TF2>RF2, for
instance, TF2=1 RF2, is realized with reference to FIGS. 6A to 6D in view
of 7A and 7B. The EPOINTER and DPOINTER, in FIG. 6A, do not start at the
same point, and the waveform shown in FIG. 7B has a corresponding shift
regarding that shown in FIG. 7A, since the transmitting frequency TF2 is
ahead of the receiving frequency RF2. Likewise, EPOINTER and DPOINTER
return to their initial state as shown in FIG. 6A, after DPOINTER travels
two rounds of the unit cycle and EPOINTER travels one round of the unit
cycle. The second situation is different from the first situation at the
point that the waveforms in FIG. 7A are shortened and repeated in the next
periods of FIG. 7B.
It should be understood that although the method of pulse code modulation
is employed in the above-mentioned drawings; i.e., FIGS. 5A, 5B, 7A and
7B, one bit encoder system (a continuous variable-slope delta modulation
(CVSD), adaptive delta modulation (ADM), or delta modulation (DM) system)
is needed to this invention to design D/A converter since there are many
discontinuities in the decoded waveforms resulted from over-writing or
over-reading. Due to the limitation of the slew rate, the one bit encoder
system cannot respond to a great voltage difference in the waveform,
therefore, the system must smooth the waveforms. Employed in this
invention, a low-pass filter in the amplifier 16 smoothes the analog
signal which is transmitted by a loudspeaker 17.
Although the DCLK and ECLK are independent of each other, some limitations
must be placed on the DCLK so that after the D/A converter 15 generates
the audio signal with some distortion, it will still be understandable,
but not identifiable. This is done by choosing appropriate time intervals
in the waveform of the input voice signal, which correspond to human voice
frequencies, and then: (1) repeating the parts of voice signal at a
certain interval; or (2) clipping some portion of the voice signal. For
example, after clipping, other portion of the voice signal is left over
and doubled in a corresponding interval, as shown in FIG. 5B.
Particularly referring to FIG. 8, for convenience of explanation, a series
of input signals is divided into several sections with waveform 1 at
interval t1, waveform 2 at interval t2, waveform 3 at interval t3,
waveform 4 at interval t4, waveform 5 at interval t5, and so on. It should
be noted that in actual situations, the waveform shown in the right side
at each interval is very similar to one other; therefore, the distortion
on the above-mentioned waveform will not make the voice signal
undistinguishable. As shown in FIG. 9, when the transmitting frequency is
at a rate equal to twice the receiving frequency, waveforms 1-4 are
respectively repeated at intervals 1-4, wherein the signal, shown in
dotted lines, output from the D/A converter is smoothed by a finite slew
rate D/A modulation system as mentioned heretofore. If the transmitting
frequency is at a rate equal to half the receiving frequency, as shown in
FIG. 10, waveforms 1-4 are cut in half and then amplified twice at
intervals 1-4, respectively; moreover, the signal output from the D/A
converter is smoothed as shown in dotted lines.
FIG. 11 is another case, in which the present invention performs the
special effect of converting an audio signal into a robot-like audio
signal. This effect is attained by repeating the waveform 1 separately at
intervals 1 to 4, and also repeating the waveform 5 separately at
intervals 5 to 8.
Referring to FIG. 12, it can be seen that another preferred embodiment of
this application is attained by employing an encoding clock ECLK2 being at
formal interval, and a pulse of another decoding clock DCLK2 being not
spaced from its precedent pulse and from its subsequent pulse by the same
time interval, so that only every other pulse of the decoding clock DCLK2
coincides with the encoding clock ECLK2. The employment of the informal
transmitting frequency of the decoding clock DCLK2 can attain the function
of voice modification; namely, the output signal is processed by
alternatively higher and lower transmitting frequences of the decoding
clock different from the receiving frequency of the encoding clock. The
coincidence of one out of two pulses of the decoding clock to respective
pulse of the encoding clock is to prevent an unrecognized output signal,
i.e., become "synchronous".
It must be pointed out that a voice modifier in accordance with the present
invention can be made according to at least three methods. The first
method is to set the transmitting frequency at a rate higher or lower than
the receiving frequency (for example, twice than the receiving frequency)
and to repeat the waveform at the same interval. The second method is to
employ an informal decoding clock, of which one out of two pulses
coincides with corresponding encoding clock. The third method is to
combine the above-mentioned first and second methods together.
The above examples and description have been given for purposes of
illustration, and are not intended to be limitative. Many variations can
be effected in the various compositions, methods and processes, without
exceeding the scope of the invention.
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