Method and apparatus for performing frequency spectrum inversion

An apparatus and method for creating voice privacy by performing frequency spectrum inversion in electronic voice transmission systems includes the steps of digitizing an analog signal and inverting the frequency spectrum of the digitized audio signal. From the inverted spectrum, a complex signal is created from which the real component is extracted to produce a real signal suitable for transmitting. The digital signal processing is performed entirely with software. The scrambling and descrambling processes are nearly identical, therefore, the same hardware and software may be used to scramble and descramble the signal. The apparatus may uses a "rolling code" to increase the effectiveness of the scrambling.

BACKGROUND OF THE INVENTION 
A. Field of the Invention 
The present invention relates to digital signal processing. More 
particularly the present invention relates to a method and apparatus for 
performing frequency spectrum inversion in electronic communications 
systems. 
B. Problems in the Art 
In the field of two-way radio communications, it is often desired to have 
secure communications between the sender and receiver. The most common 
method of providing security in two-way radio communications is by 
scrambling the transmitted audio signals and descrambling the received 
audio signals. Prior art scrambling and descrambling methods have various 
disadvantages. Most prior art devices involve hardware with excessive 
complexity and result in poor audio quality after being descrambled. 
Typical prior art designs use hardware multipliers and analog filters, for 
example. These designs cause signal loss due to the fixed bandwidth of the 
analog filter which results in poorer audio quality. Most prior art 
scrambling and descrambling systems also are inefficient and require a 
significant amount of hardware to scramble and descramble the audio 
signals. Prior art systems typically require separate scrambling and 
descrambling circuits since the scrambling and descrambling processes are 
different. 
Some prior art scrambling and descrambling methods use a "rolling code" to 
alter the scrambling method over time to reduce the chances of an 
unauthorized receiver descrambling the signals. Prior art systems using 
rolling code descramblers are limited in the frequency that the code 
changes without causing a distortion to the signal. Also, when prior art 
systems use a rolling code, spectral loss is observed. 
Therefore there is room for improvement in the art. The present invention 
represents an improvement over the state of the art. 
C. Features of the Invention 
A general feature of the present invention is the provision of a method and 
apparatus for performing frequency spectrum inversion in electronic 
communications systems which overcomes problems found in the prior art. 
A further feature of the present invention is the provision of a method and 
apparatus for performing frequency spectrum inversion in electronic 
communications systems which uses a frequency translation to shift the 
spectrum of the signal, filters out the complex baseband components, 
shifts the spectrum again using an arbitrary complex frequency shift, and 
extracts the real part of the complex signal for transmission. 
A further feature of the present invention is the provision of a method and 
apparatus for performing frequency spectrum inversion in electronic 
communications systems which eliminates the need for tunable filters and 
the like by digitizing an audio signal and uses a digital signal processor 
(DSP) to process the signal. 
A further feature of the present invention is the provision of a method and 
apparatus for performing frequency spectrum inversion in electronic 
communications systems which creates a scrambled signal which can be 
efficiently and reliably sent through wireless communication systems or 
over telephone lines. 
A further feature of the present invention is the provision of a method and 
apparatus for performing frequency spectrum inversion in electronic 
communications systems which uses software to process the digitized audio 
signal rather than discrete analog components. 
A further feature of the present invention is the provision of a method and 
apparatus for performing frequency spectrum inversion in electronic 
communications systems which preserves more of the original audio spectrum 
than previous methods. 
A further feature of the present invention is the provision of a method and 
apparatus for performing frequency spectrum inversion in electronic 
communications systems which improves the security of the transmission by 
allowing for more rapid changing of inversion frequencies than in previous 
methods. 
These as well as other objects, features, and advantages of the present 
invention will become apparent from the following specification and 
claims. 
SUMMARY OF THE INVENTION 
The present invention relates to a method and apparatus for processing 
digitized audio signals to scramble and descramble audio signals for 
providing security in electronic communications. The signals are processed 
by inverting the frequency spectrum of the digitized audio signal. From 
the inverted spectrum, a complex signal is created from which the real 
component is extracted to produce a real signal suitable for transmitting. 
The processing method may optionally include the step of varying the 
inversion frequency to provide further security. 
An apparatus for practicing the method may include an analog to digital 
converter for sampling and digitizing an audio signal, a processor for 
processing the digitized audio signal, and a digital to analog converter 
for converting the digitized processed signal to an analog signal. The 
scrambling and descrambling processes are identical, therefore, the same 
hardware and software may be used to scramble and descramble the signal.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT 
The present invention will be described as it applies to its preferred 
embodiment. It is not intended that the present invention be limited to 
the described embodiment. It is intended that the invention cover all 
alternatives, modifications, and equivalences which may be included within 
the spirit and scope of the invention. 
Generally, the present invention is a digital signal processing technique 
by which a sampled audio signal is modified (scrambled) by rearranging its 
frequency components. In this way, the content of the message of the 
transmitted signal is preserved, but the message is unintelligible unless 
processed by a descrambler when received. Audio signals in the frequency 
range of approximately 300 hertz (Hz) to 3 kilohertz (KHz) are subjected 
to the process of the present invention in which the frequency spectrum of 
the signal is inverted. This scrambling process renders the resulting 
audio signal virtually unintelligible. The scrambled signal can then be 
sent through wireless systems or over telephone lines with the content of 
the message protected. When the scrambled signal is received by the 
receiver the signal is subjected to the descrambling process to recover 
the original audio signal. 
Real signals have a property of symmetry in the frequency spectrum. Since 
the scrambling process desires a non-symmetrical spectrum, complex signals 
must be used. For real time varying signals a(t) and b(t), a complex 
signal is constructed as c(t)=a(t)+jb(t) where j=sqrt(-1), a(t)=real part; 
jb(t)=imaginary part. A(t) and b(t) can be physically processed as two 
real signals using complex operational rules. When a(t)=cos.omega.t and 
b(t)=sin.omega.t then c(t)=cos.omega.t+jsin.omega.t=e.sup.j.omega.t. 
Briefly, the digital signal processing technique used by the present 
invention begins with the analog voice signal being sampled and digitized 
using a conventional analog to digital converter. The sampled audio signal 
is then subjected to a positive complex frequency translation that centers 
the negative frequency components of the desired audio signal around 0 Hz. 
After the frequency shift, the audio signal is subjected to a low pass 
filter so that only the baseband components remain. The filtered complex 
baseband signal is then subjected to an arbitrary complex frequency shift 
to center the signal frequency in a desired frequency band. The resulting 
signal spectrum occupies roughly the same bandwidth as the original signal 
but the frequency spectrum of the signal has been inverted. The final 
audio signal is produced by extracting the real part of the complex 
samples. These samples are then processed in a digital to analog converter 
to generate an analog waveform. This waveform is a distorted version of 
the input audio signal and the message content is essentially 
unintelligible. The scrambled signal can be recovered or descrambled with 
minimal distortion by applying the described process a second time while 
reversing the order of the frequency shifts. 
FIG. 1 shows a block diagram illustrating the scrambling process by which 
the audio spectrum is inverted. The present invention performs the 
inversion entirely with software. 
FIG. 2 shows a block diagram illustrating the descrambling process by which 
the audio spectrum is inverted back to its original spectrum. The present 
invention also performs the descrambling process entirely with software. 
FIGS. 3 and 4 show more detailed block diagrams of the processes shown in 
FIGS. 1 and 2. FIGS. 3 and 4 show the method used to implement the process 
in a DSP chip. The symbols and descriptions in FIGS. 3 and 4 indicate to 
one skilled in the art the functions that the software of the present 
invention performs. 
FIGS. 6A-6E and 7A-7E are a sequence of diagrams illustrating the resulting 
signal spectrums at various stages the scrambling and descrambling process 
shown in FIGS. 1 and 2. The letters A, B, C, D, and E shown in FIG. 1 
correspond to the FIGS. 6A, 6B, 6C, 6D, and 6E, respectively, which each 
show the frequency spectrum of the signal at the stage of the process 
shown in FIG. 1. Likewise, the letters A, B, C, D, and E shown in FIG. 2 
correspond to the FIGS. 7A, 7B, 7C, 7D, and 7E, respectively, which also 
each show the frequency spectrum of the signal at the stage of the process 
shown in FIG. 2. FIGS. 6A-6E differ from FIGS. 7A-7E because different 
signals are introduced at step A and the inversion frequencies are 
switched around. FIGS. 6 and 7 show a detailed example of the scrambling 
and descrambling process for an inversion frequency of 3500 Hz using the 
relation: 
EQU f.sub.inv =f.sub.1 +f.sub.2 =1650+1850=3500 
f.sub.1 is fixed at 1650 Hz, while f.sub.2 can range from 639 Hz to 2446 
Hz. 
At the beginning of the process, the audio signal is sampled at an 8 KHz 
rate and digitized using the analog to digital converter 12 shown in FIG. 
1. The resulting signal spectrum is shown in FIG. 6A. The resulting 
spectrum includes an upper sideband 14 and a lower sideband 16. The 
digitized signal is then multiplied by a complex tone with a fixed 
frequency of 1650 Hz. This positions the negative frequency components 16 
of the original signal from -1350 Hz to 1350 Hz and the positive 
components 14 are positioned from 1950 Hz to 4000 Hz with some aliasing 
14A in to the negative frequencies. A complex tone is represented by 
e.sup.+j.omega.t =cos.omega.t+jsin.omega.t. This complex tone comes from a 
numerically controlled oscillator (NCO) 17 shown in FIGS. 1 and 2 and 
discussed in detail below. The resulting signal spectrum is shown in FIG. 
6B. This translates the lower sideband 16 (negative frequencies) of the 
audio signal (-3 KHz to -300 Hz) to DC as discussed above creating a 
complex baseband signal. The upper sideband 14 and 14A is an unwanted term 
and needs to be removed by filtering. 
By implementing a low pass filter 18 with a bandwidth of 1350 Hz, the 
undesired sideband and any components from the desired sideband that are 
above .+-.1350 Hz can be suppressed. As shown in FIGS. 1 and 2, a low pass 
filter 18 is used. The resulting signal spectrum is shown in FIG. 6C. As 
shown in FIG. 6C, the signal coming out of the low pass filter 18 contains 
only the desired sideband 16. 
The final step in the process is to apply an arbitrary complex frequency 
shift. The example shown in FIG. 6D. uses a frequency of 1850 Hz, although 
other frequencies are also used. This is accomplished with mixer 34 and 
NCO 17. The resulting signal spectrum is shown in FIG. 6D. This step 
shifts the inverted spectrum to the 500 Hz to 3200 Hz band. At this point 
the signal is still complex since its frequency spectrum is single sided, 
i.e., there is no complimentary frequency components in the negative 
frequency band. To produce a real signal (having complimentary frequency 
components in the positive and negative frequency bands) which is required 
for transmission, the real component of the complex signal is extracted 
and the imaginary component is discarded. The resulting signal spectrum is 
shown in FIG. 6E. The resulting signal has a spectrum that corresponds to 
an in-place inversion of the original audio spectrum which is shown in 
FIG. 6A. 
These processing steps must be performed on sampled data. Preferably, the 
processing is performed with software although an ASIC 
(Application-Specific Integrated Circuit) or FPGA could be used to perform 
the required operations. The preferred embodiment uses a Digital Signal 
Processor (DSP) part number TMS320C50. This DSP provides adequate 
resources and functionality in a lost cost, low part count chip set. 
Assembly programming language is used for efficiency, although other 
languages could be used. 
The resulting digitized real signal can then be converted to an analog 
signal by digital to analog converter 36 shown in FIGS. 1 and 2. The 
analog signal can then be transmitted by a wireless system or over a phone 
line, for example. The scrambled audio signal can then be received by a 
receiver and descrambled. 
The scrambling/descrambling algorithm described above is symmetric except 
for switching the inversion frequencies. In other words, the scrambling 
and descrambling algorithms use nearly identical processing. FIGS. 7A 
through 7E show the corresponding spectrums for the descrambling of the 
signal scrambled in FIGS. 6A-6E. 
Since real filters are not perfect, with real filters there is necessarily 
a transition band between the passband (the frequency band passed by the 
filter) and the stopband (the frequency stopped by the filter) of the 
frequency spectrum. Finite Impulse Response (FIR) filters are used as 
filter 18 of the present invention because of the necessity of avoiding 
phase distortion in the recovered audio signal. A practical FIR filter can 
be used since there is an allowable transition band of 600 Hz to work 
with. 
The lowpass filter 18 can be efficiently realized by using a Third Band 
Polyphase Filter. A Third Band Filter is a special type of filter that has 
two important properties. First, its bandwidth is equal to 1/3 the 
available bandwidth as determined by the sample rate (1333.3 Hz for an 8 
KHz sample rate). Second, when implemented as an FIR filter, every third 
coefficient (counting from the center peak value) is identically zero. 
This reduces the required computations by 33%. The term polyphase refers 
to a specific multi-rate implementation of an FIR filter that can be used 
when making sampling rate changes. For a factor of N sample rate change, 
the polyphase implementation saves a factor of N in computations. 
The NCOs 17 used to perform the frequency translations in the present 
invention are identical, with the exception that the NCO 17 used to do the 
arbitrary frequency shift must necessarily have a variable input for its 
frequency word. A block diagram of the NCO structure is shown in FIG. 5. 
The NCOs 17 are realized using a phase accumulator 50 that repeatedly adds 
an input frequency word, a value in an N-bit register 52, and one output 
of flip flop 54. The value in the register 52 accumulates and overflows 
continually, at a rate that depends on the input frequency word value and 
the bit width of the register. The most significant bits (MSB) of the 
number in the register are used to address a block of memory that contains 
sample values from a sinusoidal waveform. By varying the frequency word 
input, a range of frequencies can be assigned to the synthesized output 
waveform that results. 
The variables allowed in the implementation are the bit widths of the phase 
accumulator, the address and the data value. Since a digital signal 
processing (DSP) chip is used, it is logical to select the bit width of 
the data value to equal the bit width of the processor, usually 16 bits. 
This will yield a quantization noise floor of -98 dBc. The number of 
address bits depends on the spurious signal to noise ratio (SNR) required 
in the synthesized sinusoid. Ten bits of address will yield maximum spur 
levels in the synthesized output of -60 dBc and requires 1024 words of 
memory. Actually, since all the values for a sinewave can be determined 
from the first 90 degrees of the waveform, only 1/4 of a sinewave need be 
stored, which would require only 256 words of memory. For phases of the 
sinewave beyond 90 degrees, the values in memory are either addressed in 
reverse order, negated or both in order to produce the entire sinewave. 
The number of bits in the accumulator is determined by the required 
frequency accuracy. A 16 bit phase accumulator will yield a residual 
frequency error in the synthesized output of 0.366 Hz, which is adequate. 
The present invention operates as follows. A user of a communication 
system, will utilize the present invention to provide voice privacy by 
performing frequency spectrum inversion. If the invention is utilized by 
installation in a communication system, scrambling and descrambling is 
automatic. The system is installed by placing the system between the 
transceiver and the microphone or speaker such that before a signal is 
transmitted by the system it is processed (scrambled) and after a signal 
is received by the system it is processed (unscrambled). The user of 
transceiver has control of the device to enable or disable scrambled 
communications as desired by a button or keypad combination. The process 
operates as follows. First, an audio message is sampled and digitized by 
the A/D converter 12. The digitized audio signal then goes through the 
scrambling process described in detail above. The scrambled signal is 
converted to an analog audio signal and transmitted. Another user having a 
receiver receives the scrambled analog audio signal. First, the analog 
signal is digitized by the A/D converter 12. The digitized signal is then 
descrambled using the process described in detail above. The unscrambled 
signal is then converted to an analog signal and used by the second user. 
The second user can transmit a signal to the first user in the same 
manner. In doing so, the same process may be used to scramble and 
descramble the signals. Security of the system can be enhanced by making 
the arbitrary complex frequency shift vary with time (i.e. a "rolling 
code"). Using a rolling code requires that the receiver and transmitter be 
synchronized for proper signal recovery. An example of a rolling code 
which could be used with the present invention is a linear frequency sweep 
between -1333 Hz and +1333 Hz using a triangular waveform at a fixed 
frequency. The scrambling method of the present invention may be used as 
the sole scrambling method of a communications system or may be used with 
other systems such as a spectral rotation system. 
The preferred embodiment of the present invention has been set forth in the 
drawings and specification, and although specific terms are employed, 
these are used in a generic or descriptive sense only and are not used for 
purposes of limitation. Changes in the form and proportion of parts as 
well as in the substitution of equivalents are contemplated as 
circumstances may suggest or rendered expedient without departing from the 
spirit and scope of the invention as further defined in the following 
claims.