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Donald L. Hall Matthew I. McTaggart William K. Jenkins

School of Electrical Engineering School of Electrical Engineering School of Electrical Engineering

and Computer Science and Computer Science and Computer Science

Pennsylvania State University Pennsylvania State University Pennsylvania State University

University Park, PA 16802, USA University Park, PA 16802, USA University Park, PA 16802, USA

Email: dqh5265@psu.edu Email: mim5548@psu.edu Email: jenkins@engr.psu.edu

Abstract – A digital stethoscope is constructed for the into account the potential presence of noise that could

adaptive mitigation of internal and external noise resonate from the patient themselves, i.e. internal noise.

accompanying real-time acoustical heart This form of resonation can range anywhere from heavy

measurements. The developed system utilizes the breathing to yelling from a patient in pain, which is

fundamentals of statistical tendencies accompanying resonated through the channel accompanying the heart

signals for the suppression of noise, reconstruction of signal in the form of attenuation.

heart signals, and heart rate calculation in a The current digital stethoscope supports a real-time

simultaneous manner. In this paper, the designed display of the heart rate and acoustic heart signal, along

digital stethoscope is reconstructed from a with the audio playback to any speaker or listening device.

mathematical model accompanied by the physical A median filter was constructed for accurate heart rate

equipment used. After the exploration of the system realization. Four cascaded second order infinite impulse

from mathematical principles is completed, the response (IIR) transfer functions were cascaded for

working performance of the instrument is displayed generating an eighth order low-pass filter sufficient in peak

followed by the derivation and implementation of the detection of the heartbeat signals. An adaptive thresholding

variable step-size least-mean-squares adaptive noise algorithm was implemented for efficient peak detection

cancelling algorithm. capable of evaluating different users. The display and

Index Terms – Adaptive filters, variable step-size LMS audio playback of the microcontroller was designed as an

algorithm, digital filters, microcontroller, cardiac eighth order IIR bandpass filter for capturing critical

acoustics, internal noise reduction, digital stethoscope information regarding the first and second (S1 and S2)

acoustic parameters of the heartbeat waveform [5]. The

I. INTRODUCTION AND BACKGROUND design was also based on two cascaded fourth order

transfer functions. A variable step-size LMS adaptive

Digital stethoscopes are instruments used as a

algorithm was employed for internal noise suppression

measurement and analysis tool for auscultation (listening

based on [7]. A simulated adaptive noise canceler was

to the heart sounds of the heart) and medical diagnosis.

generated as a proof-of-concept.

Acquisition of signals accompanying the low signal

strength of the heart requires sensitive equipment in order

II. HARDWARE SETUP

to generate accurate calculations for the medical

examiners, as well as granting the ability to audibly discern The digital stethoscope is realized using a microphone

the true signal to a high degree. As the sensitivity of any placed at the inner portion of the resonator cavity, such that

receiver or recording device increases for increased signal resonations can occur from vibrations present at the analog

observability, so does the presence of unwanted signals. diaphragm. The microphone was biased at 3.3 volts from a

The presence of external noise can attenuate the pre- peripheral pin onboard of the LPCXpresso54608. The

existing heart signal such that observations of the heart signal is sampled on the LPCXpresso54608 platform at

signal are unobservable [1]. 8000 Hz at the audio-in port of the analog-to-digital

Adaptive filtering is an ever-lasting solution for the converter (ADC). The sampled data is collected by the

mitigation of time-variant noise that can occur in a system. LPC54608 for the display of the heart signals and heart rate

Such implementations have been developed for digital calculation. The digital stethoscope reconstructs the signal

stethoscopes in adaptively mitigating external noise [2], and allows for the audio playback on any listening device

[3], [4], [5] and [6]. Although proven to be successful in connected to the audio-out port of the microcontroller. The

the suppression of the external noise, and extremely current system is configured and visualized by figure 1.

valuable to the medical industry, the authors had not taken

𝑦1 [𝑛] = 𝛼10 𝑥[𝑛] + 𝛼11 𝑥[𝑛 − 1] + 𝛼12 𝑥[𝑛 − 2] (1)

+ 𝛽11 𝑦1 [𝑛 − 1] + 𝛽12 𝑦1 [𝑛 − 2]

+ 𝛽21 𝑦2 [𝑛 − 1] + 𝛽22 𝑦2 [𝑛 − 2]

+ 𝛽31 𝑦3 [𝑛 − 1] + 𝛽32 𝑦3 [𝑛 − 2]

+ 𝛽41 𝑦4 [𝑛 − 1] + 𝛽42 𝑦4 [𝑛 − 2]

Designed digital stethoscope. digitized heartbeat signal, and 𝛼 and 𝛽 are the impulse

coefficients of the IIR transfer function. The low-pass filter

III. CURRENT PERFORMANCE is used to filter the microphone signal so that the peaks per

heartbeat are well realized from the signal for peak

The digital stethoscope consists of four distinct sections detection and heart rate calculation.

for realizing the acoustical nature of the heartbeat from the

user. The functionality is resembled by the block diagram B. Threshold Adaptation and Median Filtering for Peak

shown below in figure 2. Detection and Heart Rate Calculation

The output of the low-pass filter is analyzed by a

thresholding algorithm, of which adapts to the signal being

observed in an iterative process. The heart rate filtered data

ranges from 1 to 32,767 as the buffer is calculated based on

the number of samples between two measured peaks. A

small threshold is set and is stepped up or down based on

the amount of peaks observed. Once a second peak is

detected, the time per heartbeat is calculated by the number

of samples between two peaks divided by the sample rate.

Fig. 2. High-level overview of the designed digital stethoscope on As each user has a different heartbeat, the threshold

the LPC54608 microcontroller. must adapt. The magnitude of the low pass stores 16 bits

per sample. The adaptive threshold starts at 3,500 and

The four primary sections encompass the peak detection, decreases by 2,000 every second if no measurements are

heart rate calculation, waveform display, and audio above the threshold. The adaptive threshold never goes

playback; of which consists of three sub-functions below 2,000. The adaptive threshold increases by 2,000 if

including: low-pass filtering for peak detection, an any measurement is above the threshold. Eventually the

adaptive thresholding algorithm for realizing a variety of threshold stabilizes such that it can be used to detect the

different users, a median filter for accurate heart rate peaks of each heartbeat. As the threshold is adaptive, it

calculation, and a bandpass filter for the audio playback does have a transient response until it begins to calculate

and waveform display. the heartrate accurately.

When attempting to calculate the heartrate, the peak

A. Low-pass Filtering detection governs the overall accuracy of the heart rate

calculation. As such, whenever a heart rate is calculated a

The designed low-pass filter is constructed as four median filter is leveraged for storing the data samples. The

cascaded, second order, transfer functions that feed a median filter is used to filter the heart rate buffer to always

decision making algorithm, such that the heart rate can be display the median value, or the current value, of the heart

calculated optimally. The low-pass filter is designed with a rate. The filter simply works such that it sorts each value in

cutoff frequency of 125 Hz. The transfer function is the odd-dimensioned buffer in ascending order, and

modeled by the difference equations expressed by displays the value in the middle of the buffer. An example

equations (1-4). The output of the IIR filter is dependent on of the sorting in heart rate buffer is represented in equation

both the input to the system as well as the previous outputs. (5), as the order of statistics (O.S) are represented by 𝑠𝑛 .

The overall filter is constructed as an eighth order IIR filter

of the form: 𝐵𝑢𝑓𝑓𝑒𝑟 → 𝑂.𝑆 [{𝑠1 }, {𝑠2 }, … {𝑠6 } … {𝑠10 }, {𝑠11 }] (5)

where

𝐻𝑒𝑎𝑟𝑡 𝑅𝑎𝑡𝑒 = {𝑠6 } (6) 𝑑(𝑛) = 𝑠(𝑛) + 𝑛(𝑛) (9)

C. Bandpass Filtering for Audio Playback and Waveform Display and the estimated noise source is modeled as:

The bandpass filter was implemented as an eighth order IIR 𝐿−1

filter comprising of two cascaded, fourth order sections centered

𝑛̂(𝑛) = ∑ 𝑤𝑘 (𝑛)𝑛̃(𝑛 − 𝑘) (10)

at 175 Hz and having a bandwidth of 250 Hz, is expressed here in

𝑘=0

the form of difference equations:

with

𝑦1 [𝑛] = 𝛼10 𝑥[𝑛] + 𝛼11 𝑥[𝑛 − 1] + 𝛼12 𝑥[𝑛 − 2] 𝑒(𝑛) = 𝑑(𝑛) − 𝑛̂(𝑛) = 𝑑(𝑛) − 𝑤(𝑛)𝑛̃(𝑛) (11)

+ 𝛼13 𝑥[𝑛 − 3] + 𝛼14 𝑥[𝑛 − 4] + 𝛽11 𝑦1 [𝑛 − 1] (7)

+ 𝛽12 𝑦1 [𝑛 − 2] + 𝛽13 𝑦1 [𝑛 − 3] + 𝛽14 𝑦1 [𝑛 − 4] where 𝑠(𝑛) is the acoustic heart signal, 𝑛(𝑛) is the internal

noise, and 𝑒(𝑛) is the error signal fed back to the adaptive

𝑦2 [𝑛] = 𝛼20 𝑦1 [𝑛] + 𝛼21 𝑦1 [𝑛 − 1] + 𝛼22 𝑦1 [𝑛 − 2] filter for updating the filter tap weights via gradient

+ 𝛼23 𝑦1 [𝑛 − 3] + 𝛼24 𝑦1 [𝑛 − 4] + 𝛽21 𝑦2 [𝑛 − 1] (8) descent:

+ 𝛽22 𝑦2 [𝑛 − 2] + 𝛽23 𝑦2 [𝑛 − 3] + 𝛽24 𝑦2 [𝑛 − 4]

𝑤(𝑛 + 1) = 𝑤(𝑛) − 𝜇∇𝑚𝑠𝑒 (𝑛) (12)

where equation (7) refers to the first fourth order section where ∇𝑚𝑠𝑒 is the gradient of the mean square error of the

and equation (8) is the second fourth order section, form:

respectively.

∇𝑚𝑠𝑒 (𝑛) = 𝑒 2 (𝑛) = [𝑑(𝑛) − 𝑛̃(𝑛)]2 (13)

IV. ADAPTIVE NOISE CANCELLING resulting in

𝐿−1 2

studies conducted by [1-6], in which the step-size is a fixed 𝑘=0

value. When the adaptation begins, and the tap weights are with

far from the optimal solution when first being adjusted, the ∂∇𝑚𝑠𝑒 (𝑛) ∂𝑒 2 (𝑛) ∂𝑒(𝑛)

= = 2𝑒(𝑛)

step-size should be large in order to move the weights more ∂𝑤𝑘 (𝑛) ∂𝑤𝑘 (𝑛) ∂𝑤𝑘 (𝑛)

rapidly to the desired solution [7]. The idea behind the (15)

LMS algorithm implemented here is to ensure proper = −2𝑒(𝑛)𝑛̃(𝑛 − 𝑘)

convergence in the mean with modest increase in

computation, which in turn results in a considerable resulting in the VSS-LMS gradient approximation for tap weight

adjustment as:

convergence rate, this is referred to as the variable step-size

LMS algorithm [7]. 𝑤(𝑛 + 1) = 𝑤(𝑛) + 2𝜇(𝑛)e(n)𝑛̃(𝑛 − 𝑘)

The adaptive noise cancelling algorithm employed with (16)

the digital stethoscope was that of the variable step-size with 𝜇(𝑛) subject to: 𝜇𝑚𝑖𝑛 < 𝜇(𝑛) < 𝜇𝑚𝑎𝑥

(VSS) least-mean-squares (LMS) algorithm. It was

constructed based on figure 3 below. The VSS-LMS algorithm defined in this project was

developed such that it adapts the step-size depending on the

difference between the input power of the desired signal

and the resulting error power at the output. If the error

signal approaches the desired signal, the difference in

signal power will approach zero. If the power of the output

error signal does not approach the power of the desired

signal, then the power of the error signal still contains

Fig. 3. Adaptive filter structure used in the VSS-LMS adaptive power associated to the noise power. Utilizing these

noise canceling procedure. underlying facts associated with the signal’s relative

where 𝑑(𝑛) is the primary diaphragm source corrupted by power, step-sizes can be directed such that optimal

some internal noise from the patient and 𝑛̃(𝑛) is a second convergence can be conducted in the VSS-LMS algorithm.

reference diaphragm placed on the patient away from the If the difference power is large, then the step-size

primary heart signal, such that the noise reference is should be larger so that the system can adapt more quickly.

considered uncorrelated to the heartbeat signal. Similarly, if the difference in power is small, then the step-

size should be small so that the system does not need to

A. VSS-LMS Algorithm adapt as quickly due to the close approximation of the

desired signal. The step-size was variably adjusted for

The input 𝑑(𝑛), i.e. the desired signal, is that of the optimal convergence based on relative power ratios

heart acoustics resonated from the resonator cavity and expressed as:

recorded by the microphone. It is modeled as:

𝑃𝑒𝑟𝑟𝑜𝑟 |𝑒(𝑛)|2 since 𝛼(𝑛) and 𝑠(𝑛) are uncorrelated processes, thus

𝑥= = (17) resulting in (22) after resubstituting 𝑛(𝑛) − 𝑛̂(𝑛) for 𝛼(𝑛):

2𝑃𝑑𝑒𝑠𝑖𝑟𝑒𝑑 2|𝑑(𝑛)|2

where step-size parameter 𝜇(𝑛), is linearly stepped as: 𝐸[𝑜𝑢𝑡𝑝𝑢𝑡 2 ] = 𝐸[𝑏(𝑛)2 ] + 𝐸[{𝑛(𝑛) − 𝑛̂(𝑛)}2 ] (23)

0.01, 𝑥 < 0.5 showing that the minimization of the noise does not affect

𝜇(𝑛) = {1.99𝑥 − 0.99, 0.5 ≤ 𝑥 ≤ 1 (18)

the signal power at the output of the adaptive VSS-LMS

1, 𝑥>1

filter. Leveraging these underlying assumptions, an

Equations (17) and (18) define the adaptive noise AWGN channel was constructed for the realization of 𝑛(𝑛)

canceling algorithm in an efficient way, such that if the and 𝑛̃(𝑛) noise references. The un-attenuated heartbeat

power of the error signal is twice the power of the desired signal s(n) is observed below.

signal, then the step-size will be set as the upper bound of

1. When the error becomes smaller and the difference

approaches zero, then the step-size reaches the lower

bound of 0.01. When the digital stethoscope is first started,

the step-size function is initialized to the lower bound,

0.01. After receiving the first x amount of samples, the

power levels are determined and the step-size is adjusted in

an iterative manner ensuring that the tap weights are

corrected effectively.

Fig. 4. Heartbeat signal to be identified by the VSS-LMS

adaptive noise canceling filter.

V. RESULTS

The signal was extracted from a database [9] and used

The VSS-LMS algorithm is an efficient algorithm for

as the heartbeat signal for simulation. Synthetic noise is

determining an optimum step-size value in the gradient

then injected into the signal with a mean of zero and a

approximation process for approximating the error. After

variance of 0.25, shown below in figure 5.

mathematically modeling the VSS-LMS adaptive noise

canceling algorithm, two simulations were conducted in

MATLAB. The first simulation encompassed the heartbeat

extraction from a pre-recorded signal embedded in additive

white Gaussian noise.

regarding the statistical nature of the additive white

Gaussian noise (AWGN) channel. When observing the Fig. 5. Additive white Gaussian noise injected into the heartbeat

adaptive filter structure of figure 3, we first must reveal that signal.

the heartbeat signal is unaffected by the error correcting

process, i.e. weight adjustment and noise cancellation. The where figure 5 depicts the desired signal 𝑛̂(𝑛) which is fed

first assumption is that the input signal 𝑠(𝑛), noise source to the VSS-LMS adaptive filter. From this, the result of the

𝑛(𝑛), reference noise source 𝑛̃(𝑛), and estimated noise VSS-LMS filter revealed an estimated heartbeat signal as

figure 6:

source 𝑛̂(𝑛) are statistically stationary with zero means [8].

The output of the adaptive filter is modeled as:

𝑜𝑢𝑡𝑝𝑢𝑡 = 𝑠(𝑛) + 𝑛(𝑛) − 𝑛̂(𝑛) (19)

which is re-written as:

𝑜𝑢𝑡𝑝𝑢𝑡 2 = [𝑠(𝑛) + 𝑛(𝑛) − 𝑛̂(𝑛)]2 (20)

substituting 𝛼(𝑛) for 𝑛(𝑛) − 𝑛̂(𝑛) and expanding gives:

𝑜𝑢𝑡𝑝𝑢𝑡 2 = 𝑠(𝑛)2 + 2𝛼(𝑛)𝑠(𝑛) + 𝛼(𝑛)2 (21)

and taking the expectation gives:

𝐸[𝑏(𝑛)2 ] + 2𝜇𝛼 𝜇𝑠 + 𝐸[𝛼(𝑛)2 ] (22)

Fig. 6. Reconstructed signal at the output of the VSS-LMS

adaptive noise canceling filter system.

B. Injected Noise Recorded by Digital Stethoscope [2] Y. Bai and C. Lu, “The Embedded Digital Stethoscope Uses

the Adaptive Noise Cancellation Filter and the Type I

The second simulation consisted of the injection of a Chebyshev IIR Bandpass Filter to Reduce the Noise of the

noise reference from a recording of a patient talking while Heart Sound.”

the diaphragm is placed on the right side of the patient’s

[3] F. Belloni, D. D. Giustina, M. Riva, and M. Malcangi, “A

chest. A second recording is made on the same patient for New Digital Stethoscope with Environmental Noise

measuring the patient’s heartbeat signal. Using these two Cancellation,” ADVANCES in MATHEMATICAL and

sources, the construction of the desired signal and noise COMPUTATIONAL METHODS.

reference was feasible. Figure 7 depicts the measured

heartbeat signal of the patient at rest. [4] S. B. Patel, T. F. Callahan, M. G. Callahan, J. T. Jones, G. P.

Graber, K. S. Foster, K. Glifort, and G. R. Wodicka, “An

Adaptive Noise Reduction Stethoscope for Auscultation in

High Noise Environments,” Acoustical Society of America,

May 1998.

[5] P. U. Kim, Y. Lee, J. H. Cho, and M. N. Kim, “Modified

Adaptive Noise Canceller with an Electrocardiogram to

Enhance Heart Sounds in the Auscultation Sounds,” The

Korean Society of Medical & Biological Engineering and

Springer 2011.

[6] B. E. Hill, “Reducing Background Noise through a

Fig. 7. Adaptive noise canceling via VSS-LMS algorithm with Stethoscope Cup using Adaptive Filters,” NASA Space

desired signal (left) and output signal (right). Grant Consortium, May 2010.

modeling. New Delhi: Wiley, 2014

The digital stethoscope simulations lay the foundation

[8] B. Widrow, J. M. McCOOL, C. S. Williams, J. R. Zeidler, J.

towards future growth of the VSS-LMS algorithm for

R. Glover, J. Kaunitz, R. H. Hearn, and R. C. Goodlin,

adaptive noise canceling in the medical industry. The VSS- “Adaptive noise cancelling: Principles and

LMS algorithm has proven to be an effective method at Applications,” Adaptive noise cancelling: Principles and

reconstructing the heartbeat signal, as well as mitigating applications - IEEE Journals & Magazine. [Online].

noise from internal resonations from a patient. The entire Available:

idea behind the developed digital stethoscope on the http://ieeexplore.ieee.org/stamp/stamp.jsp?arnumber=1451965

LPCXpresso54608 board is to explore the idea of creating Accessed: 07-Oct-2017].

a new method of helping medical examiners in the

diagnostic process of cardiac illness in patients. [9] “heart beat increasing 116642.wav by

Conventional methods of the LMS algorithm expressed klankbeeld,” Freesound.org. [Online]. Available:

https://freesound.org/people/klankbeeld/sounds/181805/.

highly accurate methods of adaptive noise canceling for

[Accessed: 08-Oct-2017].

digital stethoscopes, however, when noise variance is high

the step-size cannot adjust itself in helping speed up the

convergence process. Moreover, the conventional LMS

algorithm was employed for adaptively canceling the

external/background noise that can be received by the

microphone embedded within the resonator cavity of the

diaphragm connected to the traditional analog stethoscope,

which is not always going to be the source of noise.

Therefore, the implementation of the VSS-LMS algorithm,

along with the analysis of internal noise that can be present

in the diaphragm, brings a new method of adaptive noise

canceling to fruition that could be present in designed

digital stethoscopes on the market today.

VII. REFERENCES

[1] N. Jatupaiboon, S. Pan-Ngum, and P. Israsena, “Electronic

stethoscope prototype with adaptive noise

cancellation,” 2010 Eighth International Conference on ICT

and Knowledge Engineering, 2010.

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