Electronic stethoscope system and method

A microprocessor based sound enhancement system in which frequencies outside of the auditory range of the human ear are translated into sound within the auditory range by translating each frequency component of the entire frequency spectrum of the input signal by a time scale compression factor. Preferably, a microprocessor transforms an electrical signal corresponding to the input signal into a frequency spectrum signal comprising frequency components having frequency, phase, and amplitude elements by performing a fast Fourier transform (FFT) operation on the input signal. The frequency components of this transformed signal are translated and the resulting translated frequency spectrum signal is transformed into a time varying output signal by performing an inverse FFT operation on the translated frequency spectrum signal. When heart pulses or similar periodic waveforms are monitored, the pulse rate of the signal is maintained in the output signal. Alternatively, the time varying signal is compressed in the time scale and then transformed into audible sound, while maintaining the original pulse rate in the output signal, thereby resulting in the same output signal as that resulting from translating the entire frequency spectrum by the time scale compression factor.

FIELD OF INVENTION 
The present invention relates to the enhancement of sound having frequency 
components that are not within the auditory range of the human ear so that 
these frequency components become audible without altering the tonal 
relationship among these frequency components. 
BACKGROUND OF THE INVENTION 
Since the nineteenth century, it has been well known by physicians that a 
medical diagnosis can be determined by listening to the sounds emanating 
from a patient's internal organs either in operation or as a result of a 
physician's rapping on the patient's chest with his fingers. This method 
of medical diagnosis is known as auscultation. In order to amplify these 
sounds, the stethoscope was invented in 1816 by the French physician Rene 
Laennec and has since proven to be an invaluable tool in noninvasive 
medical diagnosis. 
Auscultation is considered by many cardiological authorities to be the most 
sensitive test of the functional integrity of the heart. Not infrequently, 
murmurs or changes in the heart sounds are the only concrete indication of 
heart disease. The ability to distinguish between what is normal and what 
is abnormal is particularly difficult in many borderline cases. 
Cardiovascular sounds, however, include components that have frequencies 
and intensities which are not within the auditory range of the human ear. 
As a result, only a small fraction of these sounds or acoustic vibrations 
can be heard using a conventional stethoscope. There have been attempts to 
solve the problem of low signal levels by the use of electronic 
stethoscopes using microphones and signal amplifiers. However, the use of 
such equipment is limited in frequency due to inherent insensitivity o the 
human ear, as well as the earphones used in these electronic stethoscopes. 
Electronic stethoscopes have been devised which enable a user to hear heart 
sounds more easily. For example, U.S. Pat. No. 4,528,689 discloses an 
electronic stethoscope which provides a signal which is similar to but not 
exactly the same as a slowed down version of the original sound by 
producing an output signal which is composed of replicated sets of cycles 
of the original sound. The output signal, however, does not maintain the 
tonal relationships of the original sounds, is not a truly frequency 
shifted version of the original sound, and is merely the original signal 
periodically reproduced so as effectively to have a longer duration. 
Other electronic stethoscopes have been devised which shift low frequency 
signals into a frequency within the ear's audible range, however, such 
prior art devices all have their limitations. 
For example, U.S. Pat. No. 3,562,428 discloses an electronic stethoscope 
which modulates the detected sound so that low frequency signals are 
shifted into the audible frequency range. The quality of the sound 
generated by this stethoscope is substantially different from the usual 
sounds heard by the physician since a carrier signal is added to the 
originally detected sound whereby the detected sound is merely shifted 
into a higher frequency range without maintaining the same tonal 
relationships as in the originally detected sound. 
U.S. Pat. No. 4,220,160 is directed to a device for transposing heart 
sounds into the audible frequency range by processing certain 
predetermined frequency components of the detected sound to obtain an 
output signal which consists of the sum and difference of the frequency 
components and a carrier frequency signal. The output signal is then 
filtered to remove the difference between the frequency components. 
U.S. Pat. No. 4,594,731 discloses an electronic stethoscope in which the 
detected heart sounds are frequency multiplied to transpose the sounds to 
a more favorable position within the auditory range. The signal is input 
to a frequency doubling circuit consisting of a full wave rectifier, 
bandpass filter and a wave shaping circuit. 
The main disadvantage to these prior art electronic stethoscopes is that 
the tonal relationships in the sounds generated by the electronic 
stethoscopes are different from the tonal relationships in the original 
sounds and thus do not blend in the same way. For example, a group of 
tones that are separated by octaves (Two signals are said to be separated 
by an octave if the ratio of their frequencies is equal to two.) in the 
original sound (10 Hz, 20 Hz, and 40 Hz) would no longer be separated by 
octaves in the signal after processing by the electronic stethoscopes 
because the detected signal is merely shifted to a different frequency 
range (110 Hz, 120 Hz, and 140 Hz). Thus the sounds generated by such 
electronic stethoscopes are not similar to the original sounds, do not 
combine with each other in the same way as the original sounds, and are 
unrecognizable to experienced physicians trained to recognize the sounds 
of a conventional stethoscope. Similar problems arise using the electronic 
stethoscopes in which the frequencies of the sound are simply multiplied 
by a constant value. It is important that not only the relative frequency 
relationships between signal components be maintained, but also the 
relative phase relationships as well as the amplitude of the signal 
components. The frequency multiplication circuitry described in U.S. Pat. 
No. 4,598,731 not only does not maintain the relative phase relationships 
of the frequency components, as well as the amplitudes but also generates 
new unwanted frequency components. This combination changes the shape of 
the waveform and the melody of the heart sounds so that they are no longer 
recognizable as a transposed version of the original heart sounds. 
SUMMARY OF THE INVENTION 
In the microprocessor based electronic stethoscope system of the present 
invention, heart sounds with frequencies that are not within the auditory 
range of the human ear are detected and transformed into an electrical 
input signal. The time varying signal is compressed in the time scale 
while maintaining the original pulse rate whereby the relative frequency 
and phase relationships of the signal are preserved. The output signal is 
transformed into an audible sound that has the same melody as the original 
heart sound, but at a higher pitch and includes those frequencies which 
were previously inaudible to the human ear. 
In one embodiment of the present invention the microprocessor transforms 
the electrical input signal into a frequency spectrum signal by performing 
a fast Fourier transform (FFT) on the signal. Each component of the 
spectrum is then translated in frequency by a constant time scale factor 
whereby the original inter-component frequency and phase relationships of 
the signal are maintained. An inverse FFT operation is used to translate 
the frequency spectrum signal into a time varying signal which is 
subsequently transformed into audible sound. 
In another embodiment of the present invention the input electrical signal 
is sampled at a first sampling rate and is output at a second faster 
sampling rate to translate the inaudible sounds into the audible frequency 
range. 
In the present invention, sounds having frequencies outside the auditory 
range of the human ear are brought into the auditory range without 
changing the tonal relationship among the frequency components of the 
sounds. In other words, the translated sounds maintain the melody of the 
original heart sounds but at a higher pitch to include those sounds which 
were originally inaudible. 
By using the system of the present invention to translate inaudible sounds 
into the audible frequency range, the tonal relationships of the sounds 
remain unchanged because amplitudes as well as the relative frequency and 
phase relationships of the frequency components of the signal are 
maintained. For example, a group of tones separated by octaves (10 Hz, 20 
Hz, and 40 Hz) when time compressed in accordance with the present 
invention results in tones (100 Hz, 200 Hz, and 400 Hz) which remain 
separated by octaves and have the same relative phase relationships. This 
output sound is similar to the original sound since the new tones combine 
with each other in the same way as the original sounds and thus remain 
recognizable to physicians experienced in using conventional stethoscopes. 
By choosing the appropriate time compression factor, sounds having 
frequencies above the auditory range as well as sounds having frequencies 
below the auditory range can be translated into the auditory range. 
Although the sound enhancement system of the present invention is designed 
primarily for the detection and analysis of cardiovascular sounds, other 
sounds generated by the human body can also be enhanced. Among the sounds 
that can be monitored are bowel sounds as in the diagnosis of acute 
appendicitis, brachial artery sounds, carotid artery sounds, auscultation 
of the bladder for determining urinary retention, respiratory sounds, and 
muscle sounds. 
The power requirements and size of the components used in the electronic 
stethoscope system of the present invention allow the entire system to be 
battery operated and worn on the belt as a conventional transistorized 
radio. 
If size and power requirements are not important factors, additional 
optional features can be added to the system. For example, a pulse 
separating gate can be added so that the input signal is analyzed only 
periodically. This gate is useful when it is desirable to analyze only 
particular periodic sounds, for example one of the four heart sounds. A 
signal enhancer can be added which enhances particular frequencies or 
assigns special chords, notes or note combinations to particular 
frequencies in order to produce a unique and distinctive set of sounds in 
the auditory range which easily indicate different kinds of pathology. 
Signal filters can be used to remove those frequencies which are 
undesirable. 
The system can be used in conjunction with a device which obtains an 
electrocardiogram (ECG). The time varying waveform comprising the ECG is 
related to the electric fields generated in the body due to the spread of 
electrical excitation connected with the contraction cycle of the heart. 
The ECG waveform is normally interpreted by viewing the pattern on a paper 
chart. The system of the present invention can be used in conjunction with 
the normal ECG paper chart output by processing the ECG waveform to 
provide an audio signal corresponding to the ECG waveform in which 
frequencies not within the auditory range are translated into the auditory 
range. 
The system can similarly be used with other waveforms including 
electroencephalogram (EEG) waveforms which are commonly used to diagnose 
brain and nervous system disorders, and electromyogram (EMG) waveforms 
which are used to assist patients with spinal chord or other nervous 
system disability to relearn motor control of their limbs. 
Alternatively, a number of waveforms can simultaneously be applied to the 
system of the present invention with the translated version of each 
waveform being filtered so as to have frequencies within specified 
frequency ranges or alternatively each translated waveform can be assigned 
a particular chord, set of notes or note combinations. 
The disclosed system can be connected to an additional information 
processor which, by using appropriate algorithms and stored data, will 
automatically interpret the enhanced sound to perform a medical diagnosis 
which can be displayed by a printed or video output or by the illumination 
of an appropriate indicator light. 
Although the electronic stethoscope of the present invention is designed 
primarily to aid in medical diagnosis, it can also have non-medical uses 
such as, for example, a recreational device where users of the sound 
enhancement system can listen to musical tones or chords which correspond 
to their own bodily functions. In addition, the system can be used as an 
aid in other technical fields to enhance low or high frequency signals, 
for example, in testing the structural integrity of buildings and bridges 
and in detecting or predicting earthquakes.

DESCRIPTION OF THE PREFERRED EMBODIMENT 
FIG. 1 is a schematic diagram of a specific embodiment of the present 
invention where sound detected by a sound detector is electronically 
amplified and fed into a microprocessor where the frequency components of 
the sound are processed and then selectively enhanced. This processed 
signal is then filtered and converted into audible sound. 
Referring to FIG. 1, sound detector 10 detects sound or acoustic vibrations 
emanating from a patient and converts the sound into an electrical signal. 
Sound detector 10 has a sensitivity such that it is able to detect sounds 
or acoustic vibrations having frequencies and intensities which are not 
within the auditory range of the human ear. Generally, the ear can hear 
sounds with frequencies as low as 20 Hz and as high as 20,000 Hz although 
at these extreme limits, the intensity range perceptible as sound is very 
small. A graphical representation of the auditory range of the human ear 
is shown in P.M. Morse, Vibration And Sound, Acoustical Society of 
America, 1976, page 227. Illustratively, sound detector 10 is a highly 
sensitive microphone or a stethoscope bell connected to an electronic 
circuit which converts the output of the stethoscope bell into an input 
electrical signal. The input electrical signal generated by sound detector 
10 is amplified by amplifier 12 so that the entire frequency range of the 
detected sound is amplified. 
The output of amplifier 12 is then optionally fed to gate 14 which 
periodically connects amplifier 12 to microprocessor 16. The amount of 
time T.sub.1 that gate 14 connects amplifier 12 to microprocessor 16 as 
well as the time T.sub.2 between these connections is preferably manually 
set by the operator. For example, when the electronic stethoscope system 
is used to monitor heart sounds, the gate connection period T.sub.1 and 
separation time T.sub.2 can be synchronized to the duration and beat 
frequency of a particular heart sound. The heart usually generates a 
series of two heart sounds although it may generate four sounds. The first 
sound occurs at the beginning of ventricular systole, its timing being 
revealed by the fact that it slightly precedes the rise of the carotid 
pulse. The second sound is a short sound and closely follows the closure 
of the two semilunar valves. The third sound occurs in early diastole, 
apparently at the time in which ventricular volume and compliance 
equilibrate. The fourth heart sound (presystolic or atrial sound) is 
related to atrial contraction and usually precedes the first heart sound 
but may occur in middiastole. A detailed discussion of these heart sounds 
is found in A.A. Luisada and G.S. Sainani; A Primer of Cardiac Diagnosis; 
W. H. Green, Inc.; 1968; pages 34-64. The waveform corresponding to each 
heart sound comprises a wave packet which is repeated with a period 
corresponding to the pulse rate of the heart. 
Referring to the operation of gate 14 of FIG. 1, the connection period 
T.sub.1 and separation time T.sub.2 can illustratively be selected so that 
only one of the heart sounds is detected. The output of gate 14 is thus in 
the form of a series of wave packets having a duration of T.sub.1 seconds 
and separated by T.sub.2 seconds as shown in FIG. 2. When gate 14 is used, 
single pole double throw switch 15 is provided between gate 14 and 
microprocessor 16. When the operator of the system is selecting gate 
connection period T.sub.1 and separation time T.sub.2, switch 15 connects 
gate 14 directly to sound generator 22, thereby by-passing microprocessor 
16, so that the operator can determine when the correct values of T.sub.1 
and T.sub.2 have been selected. Once T.sub.1 and T.sub.2 have been 
appropriately selected, switch 15 is switched so that gate 14 is connected 
to microprocessor 16. Thus, gate 14 periodically removes at least one wave 
packet (or heart sound) from the signal generated by amplifier 12. 
Although gate 14 is shown as separate from microprocessor 16, the 
operation performed by gate 14 can alternatively be performed by 
microprocessor 16. If gate 15 is not used, the output of amplifier 12 is 
optionally fed to a sample and hold circuit 17 and then to microprocessor 
16. 
The microprocessor 16 performs a number of mathematical operations on the 
signal generated by gate 14 to translate or time compress signals so that 
the inaudible sounds are translated into the auditory range. Referring to 
FIG. 1 and the flow chart in FIG. 3, the microprocessor first converts the 
input analog signal to a digital signal in block 23 at a predetermined 
sampling rate of 1/dt per second. Optionally the sampling may be done 
external to the microprocessor 16 by sample and hold circuit 17. Each 
sample is stored in memory as indicated by block 25. In block 26, the 
stored sample of the input signal is read from memory and output at a rate 
K times faster then the original sampling rate, where K is defined as the 
time scale compression factor. The original pulse rate is maintained by 
synchronizing the start of the output signal with the pulse rate. 
Preferably this is accomplished by an external pulse rate trigger 18 that 
may be manually adjusted to equal the pulse rate. The output signal is 
converted to an analog signal in block 27, and transmitted to the sound 
generator 22 for conversion of the electric signal into audible sounds. 
The sounds output from the sound generator maintain the same tonal 
relationships as the original heart sounds except at a higher pitch. A 
physician using the electronic stethoscope of this invention will 
recognize the different "melodies" heart sounds as well as hear those 
heart sounds which were previously inaudible. 
Alternatively the analog to digital and digital to analog signal 
conversions may be performed external to the microprocessor 16. 
Illustratively, the microprocessor 16 is a CMOS microprocessor such as the 
8 bit MC68HCllA4 microprocessor manufactured by Motorola Inc. of 
Schammburg, Ill., although other devices with similar capabilities can be 
used. 
Another embodiment of the sound enhancement system of the present invention 
is illustrated in FIG. 4. The sound detected by a sound detector 10 is 
electronically amplified by amplifier 12 add fed into a microprocessor 16 
where the signal is time compressed. The processed signal is then 
converted into audible sound. 
As discussed in detail below with reference to the flow diagrams shown in 
FIGS. 5a and 5b, microprocessor 16 performs a number of mathematical 
operations on the signal generated by gate 14 to translate signals 
corresponding to inaudible sounds into the auditory range. Among the 
operations that microprocessor 16 performs are analog to digital 
conversion of the signal, signal sampling, Fourier transform, frequency 
translation, inverse Fourier transform, digital to analog conversion, and 
storage of information in temporary and permanent memory. Alternatively, 
analog to digital and digital to analog signal conversion can be performed 
external to microprocessor 16. 
Optionally the output signal may be further enhanced. This further signal 
enhancement is preferably performed digitally by microprocessor 16 after 
the waveform has been translated in frequency and phase but before an 
inverse Fourier transform operation has been performed upon the translated 
signal. Such signal enhancement enhances the translated frequency signal 
either by amplifying particular frequency ranges or by assigning special 
chords, notes or note combinations to particular frequencies in order to 
produce a unique and distinctive output sound in the auditory range which 
can be correlated with standards of normal heart function or different 
kinds of pathology in heart function These unique sounds make the sound 
patterns which are related to various kinds of heart pathology more 
distinctive to the ear of the operator and hence easier to interpret. 
Although signal enhancement is preferably performed by microprocessor 16, 
the signal enhancement can alternatively be performed by signal enhancer 
19 (shown in dashed lines) which receives the output electrical signal 
generated by microprocessor 16. 
Similarly, the signal after frequency translation can be filtered digitally 
by microprocessor 16 to filter out those frequencies of sound that are 
undesirable. If the signal is to be filtered, this operation is preferably 
performed upon the translated signal while it is in the frequency domain 
prior to performing the inverse Fourier transform operation. 
Alternatively, signal filtering can be accomplished by analog filter 21 
which receives the output of signal enhancer 19, or alternatively, 
microprocessor 16 if signal enhancer 19 is not used. 
The output of filter 21 is used to drive sound generator 22 which produces 
sound or output vibrational waveform audible to the operator in response 
to the electrical signal from filter 21. Illustratively, sound generator 
22 is a pair of earphones. 
FIGS. 5a and 5b together show a preferred flow diagram describing the 
operation of microprocessor 16 of the embodiment of the present invention 
shown in FIG. 4. Referring to FIG. 5a, at block 30 the output generated by 
gate 14 is received by microprocessor 16. As previously discussed, when 
gate 14 is used, this signal is a series of wave packets separated by 
separation time periods as shown in FIG. 2. However, when gate 14 is not 
used and the sounds generated by the heart are being monitored, this 
signal is a series of wave packets not clearly separated by separation 
time periods. In block 32 this signal is detected by microprocessor 16 and 
converted from an analog signal to a digital signal. 
In block 34, the beginning of a heart pulse is determined. Determining when 
the beginning of a heart pulse occurs can be performed manually by the 
operator by using a threshold detector with an appropriately set trigger 
level and by using a time delay generator. The trigger level of the 
threshold detector is manually set by the operator. In addition, the 
operator initially manually activates the threshold detector when the 
operator detects the beginning of a first heart pulse. Whenever the 
trigger level set by the operator is exceeded, a trigger pulse is 
generated and the time delay generator is triggered which generates a 
delay pulse which has a duration approximately equal to the expected time 
between heart pulses. Illustratively, the duration of the time delay pulse 
is about 750 milliseconds. The time delay pulse is used to inhibit the 
operation of the threshold detector As a result, the retriggering of the 
threshold detector is automatically prevented until the next heart pulse 
occurs. The duration of the delay pulse can alternatively be adjusted 
manually by the operator to accommodate different pulse rates. 
Illustratively, the time delay generator is a Schmitt trigger with a time 
delayed reset. 
Alternatively, the beginning of heart pulses can be determined 
automatically. A peak detector is used to determine the maximum amplitude 
of the heart pulse waveforms. The trigger level of the threshold detector 
is then automatically set an arbitrary amount below this maximum 
amplitude. As is the case with manual operation, the operator must 
initially activate the threshold detector when the operator detects the 
beginning of a first heart pulse. The duration of the delay pulse produced 
by the time delay generator is determined automatically by determining the 
pulse rate period and setting the duration of the time delay pulse equal 
to an arbitrary amount below the pulse rate period. Appropriate use of 
discrete logic gates ensures that the threshold detector and time delay 
generator operate at the appropriate times. Thus, every time another heart 
pulse or wave packet is detected, another trigger pulse is automatically 
generated and a heart pulse counter is incremented by one. 
In block 36, it is determined if a predetermined number N of heart pulses 
or wave packets, corresponding to the number of trigger pulses generated 
by the pulse generating means, have been detected by the heart pulse 
counter. If N pulses have not been detected, in block 38 it is determined 
whether the input electrical signal was sampled (in block 40), within the 
last dT seconds. If the previous pulse waveform sampling occurred within 
the last dT seconds, the sampling of the waveform is delayed a fixed 
period of time as shown by block 42. If the previous pulse waveform 
sampling occurred more than dT seconds previously, the input electrical 
signal is sampled as shown by block 40 and the amplitude of the sampled 
waveform is stored in the memory portion of microprocessor 16. As a result 
of the sampling loop of blocks 36, 38, 40 and 42, the input electrical 
signal comprising the N pulses is sampled every 1/dT seconds as shown by 
the dots in FIG. 7a and a sampled input electrical signal is obtained. 
Illustratively, dT is equal to 50 microseconds although other values for 
dT can be chosen. It is then redetermined in block 36 whether N pulses 
have been detected. 
After N waveforms have been detected, a signal corresponding to the 
frequency spectrum of these sampled waveforms is obtained in block 44 as 
shown in FIG. 4b and stored in the memory of microprocessor 16. The 
frequency spectrum is preferably approximated by calculating the discrete 
Fourier transform (DFT) of the waveforms which describes the signal in 
real and imaginary components thus describing the frequency and phase of 
each component in the spectrum. This is accomplished by an algorithm 
generally known as a fast Fourier transform (FFT) which rapidly makes the 
computations required to obtain the DFT of an input signal. As a result of 
the FFT calculations, the continuous frequency spectrum of the input 
waveform signal is approximated by a series of points at different 
discrete frequencies having amplitudes which correspond to the amplitudes 
of the frequency spectrum at the different discrete frequencies. The 
sampled frequency spectrum signal obtained using the FFT operation is 
easily manipulated by performing mathematical operations on these 
frequency, phase and amplitude values as discussed in detail below. 
Since an FFT approximation comprises both real and imaginary elements, thus 
taking into account both the frequency and phase elements of the signal, 
the operation of shifting the frequency spectrum up does not alter the 
phase and frequency relationships among the components in the spectrum. 
Therefore, the compressed waveform in the time domain maintains its 
original shape. 
In order to translate components of sounds at or below the threshold of the 
auditory range of the human ear into the auditory range, according to the 
present invention, the frequencies and phase angles of these components 
are translated into the auditory range by time scale compression. Time 
scale compression is achieved by multiplying each of the discrete FFT 
values by a time scale compression factor. This time scale compression 
factor is chosen so that lower frequency components are translated to 
higher frequencies within the auditory range while other higher 
frequencies are translated so that they remain in the auditory range. The 
time scale compression factor can be preselected as indicated by block 46 
or, alternatively, can be calculated, as shown in block 48 and block 50, 
based upon the lowest frequency component detected. 
In block 48 the lowest (or base) frequency, F, is determined. The time 
scale compression factor, K, is determined in block 50 by dividing a 
standard frequency value, f, by the base frequency F. Thus, K=f/F. The 
standard frequency value f is chosen based upon an expected value for the 
base frequency so that the translated frequency spectrum remains within 
the auditory frequency range. Illustratively, the standard frequency value 
f is equal to about 15 Hz to about 30 Hz. 
In block 52, each component of the sampled frequency spectrum signal 
obtained in block 44 is multiplied by the time scale compression factor K 
determined in block 46 or 50. As a result, each discrete FFT component 
detected is translated to a higher frequency by the same factor. This time 
scale compression or translation is shown in FIGS. 6a and 6b in which a 
simplified discrete frequency spectrum of the original waveform is shown 
in FIG. 6a while FIG. 6b shows the translated frequency spectrum of that 
shown in FIG. 6a wherein the amplitudes and relative inter-component 
frequency and phase relationships are maintained. Although the time scale 
compression factor K used to obtain the frequency spectrum of FIG. 6b from 
that shown in FIG. 6a is equal to 15, this value for K is only used as an 
example and other values for K can be used. 
By using a time scale compression factor to uniformly translate the 
frequency and phase components of the original signal while maintaining 
the signal amplitude, the tonal relationship between the components 
remains unchanged. As shown in FIG. 6a, component 73 (50 Hz), component 74 
(100 Hz) and component 75 (200 Hz) are separated by octaves. As shown in 
FIG. 6b their translated components, component 83 (750 Hz), component 84 
(1500 Hz) and component 85 (3000 Hz), are also separated by octaves. 
The relative inter-component phase relationship also remains constant. 
Thus, the tonal relationships in the original sound are maintained in the 
translated sound while the entire frequency spectrum is translated into 
the audible range. 
At this point in the processing of the signal, as previously discussed with 
reference to FIG. 1, signal enhancement can optionally be performed 
digitally by microprocessor 16. Additionally, at this point the sampled 
translated frequency spectrum signal can be digitally filtered by 
microprocessor 16. 
Returning to FIG. 5b, after the frequencies of the sampled frequency 
spectrum signal are translated in block 52, a time varying output 
electrical signal is obtained in block 54 by performing an inverse FFT 
operation on the translated signal. 
The operations performed by microprocessor 16 can be described 
mathematically. If the original time varying input signal is f(t), its 
Fourier transform frequency spectrum shown in FIG. 6a is F(.omega.), and 
the translated frequency spectrum signal show in FIG. 6b is F'(.omega.), 
then 
EQU F'(.omega.)=F(.omega./K) (1) 
where K is the time scale compression factor. As is known from algebra, in 
order to translate the frequency spectrum F(.omega.) by a time scale 
compression factor K to obtain F'(.omega.), the frequency must be divided 
by the K as shown in equation (1) above. 
By using the inverse Fourier transform operation, the time varying output 
signal f,(t) corresponding to the translated frequency spectrum signal 
F'(.omega.) is therefore: 
##EQU1## 
substituting equation (1) into equation (2) results in the following: 
##EQU2## 
If .omega./K=.omega.', then .omega.=K.omega.', d.omega.=Kd.omega., and 
##EQU3## 
Taking the inverse Fourier transform of the right side of equation (4) 
results in the following: 
EQU f'(t)=Kf(Kt) (5) 
Therefore, when each Fourier transform of the original signal is translated 
by multiplying each component by the time scale compression factor K, and 
if K is greater than one, the resulting varying output signal is 
equivalent to the time varying input signal compressed in time by a factor 
of K. 
Preferably, the time varying output signal is then analyzed by 
microprocessor 16 to determine when heart pulses occur in the time scale 
compressed time varying output signal. The beginning of each pulse is then 
separated in time from the previous and next pulse to re-establish the 
timing of pulses present in the original time varying signal prior to time 
scale compression and processing. Alternatively, the timing of the heart 
pulses can be detected and the timing can be re-established by a means 
other than microprocessor 16, for example by the use of threshold 
detectors, time delay generators and discrete logic as discussed 
previously. 
Returning to FIG. 5b, the time varying time scale compressed output signal 
is then optionally repeated a number of times as shown in block 56. At 
block 58 the signal is converted from a digital signal to an analog signal 
and is transmitted to signal enhancer 19, filter 21, and sound generator 
22 shown in FIG. 4. 
As mathematically shown above, translating low frequency components of a 
time varying signal into higher frequency range by a time scale 
compression factor maintains the tonal relationships in the original 
signal and is equivalent to compressing in time the original signal by a 
fixed factor. This is illustrated in FIGS. 7a and 7b in which 7a is the 
time varying input waveform and 7b is the time varying output waveform 
compressed in the time scale by a factor K, thereby maintaining the signal 
amplitude and relative frequency and phase relationships of the signal. 
While the invention has been described in conjunction with specific 
embodiments, it is evident that numerous alternatives, modifications, 
variations, and uses will be apparent to those skilled in the art in light 
of the foregoing description.