Electronic stethoscope convert acoustic sound waves

Electronic stethoscope convert acoustic sound waves

1.1 Project Objective

The project objective is to design and construct an electronic stethoscope to convert acoustic sound waves to electrical signals which can then be amplified and processed for analysis.

The specific objectives of this project include, but are not limited to:

  • Conversion of acoustic sound waves to electrical signals
  • Pre-processing of auscultation signals, involving removal of interference noise and separation of respiratory noises, and amplification of the signals
  • Transmitting the heart sounds signal to a personal computer using a graphical user interface for analysis and to build-up the data-base for replay, demonstration or consultation purposes
  • Calculation and display of heart beat rate

1.2 Overall Objective

Recognition of heart disease is an important goal in medicine. The majority of stethoscopes currently on the market are acoustic devices that use purely passive mechanical parts to isolate and focus sound generated by the body. Unfortunately, though these methods have been used for years, the simplicity of such devices is overshadowed by poor sound quality. These devices are also difficult to interface with modern technologies such as computers to record and analyze body sounds. The goal of this project is to design and construct an electronic stethoscope that is comparable in cost, has better acoustic response, and can interface with modern technologies better than the current acoustic stethoscope.

1.3 Proposed Approach and Methodology

In order to extract reliable diagnostic features from heart sound it is important to first suppress noise. During heart sound acquisition many external body noises such as ambient noise, as well as internal body noises such as heavy breathing, speech and etc., may be captured. These noises are mixed with heart sound.

The approach to this project is to keep things as simple as possible. The stethoscope system detecting heart sounds has two main blocks: Signal Acquisition and Pre­processing (Signal Conditioner), and Analysis of heart sound. It consists of two parts: Hardware and software. The hardware component consists of a power supply, sensor, preamplifier, low pass filter and power amplifier. The software component consists of a segmentation and analysis algorithm. The block diagram below (Figure 1.1) shows the key parts required in this project.

1.3.1. Signal Acquisition and Pre-Processing (Signal Conditioner)

The circuit will be operating on its own portable power source. The power source and circuitry will provide adequate voltage for the electronic components to function properly. There will be a sensor to convert the acoustic signal from the stethoscope into electrical signal. The signal will be pre-amplified using op-amps. A low pass filter will be designed to remove interference noise. Finally, the filtered signal will then be amplified again using power amplifiers.

1.3.2. Heart Sound Analysis

The signal is processed through analysis algorithm before a final diagnosis can be made. The analysis algorithm is based on the analysis of heart sounds and separates the heart sound into individual cycles with each cycle containing First Heart Sound (S1), Systolic Period, Second Heart Sound (S2) and Diastolic. The main idea is that first the location of S1 and S2 will be computed and then based on that information the location of systolic and diastolic periods will be calculated. This allows further analysis on heart sounds and murmurs. The algorithms will be designed and implemented in Matlab.



The term Auscultation means, literally, ‘to listen' is the act of listening to the sounds made by internal organs, is a valuable medical diagnostic tool. Auscultation methods provide the information about a vast variety of internal body sounds originated from the heart, lungs, bowel and vascular disorders. The ability to detect heart sounds may be influenced by a number of factors, including the presence of ambient noise or other sensory stimuli. Therefore, only the skilled physician who is proficient in the skill of auscultation is likely to make accurate diagnoses upon cardiac auscultation. This may result in inaccurate or insufficient information due to the inability of the user to discern certain complex, low-level, short duration or rarely encountered abnormal sounds. It is, thus, desirable to enhance the diagnostic ability by processing the auscultation signals electronically and providing a visual display and automatic analysis to the physician for a better comparative study.

2.1 Background to the Project

Stethoscopes are medical instruments that help doctors detect sounds produced within the body, in particular heart and lungs, for monitoring the physiological condition of a patient. They typically include diaphragm detectors for picking-up sounds of lower frequency, and bell-shaped detectors for picking-up sounds of higher frequency.

Typically, heart sounds and murmurs are of relatively low intensity and are band limited to about 10-1000 Hz. Meanwhile, Speech signal is perceptible to the human hearing.

Therefore, auscultation with an acoustic stethoscope is quite difficult. Only a small proportion of cardiovascular sound energy is audible by the human ear, see Figure 2.1, [2]. The problem with acoustic stethoscopes is that the sound level is low, making diagnosis more difficult. By amplifying body sounds, electronic stethoscopes overcome the low sound levels. Modern electronic stethoscopes include electrical transducers for converting the sounds to electrical signals and amplification of the signals.

2.2 Existing Work Related to the Project

There are several commercially available electronic stethoscopes on the market. One of them is the Littmann Electronic Stethoscope Model 3000 manufactured by 3M [3]. This stethoscope is designed with a high level of amplification of up to 18 times greater than the best non-electronic stethoscopes. It features an ergonomic design and simple push button control to change modes without interrupting auscultation. Its' new Ambient Noise Reduction technology reduces distracting room noise by an average of 75% without 5 eliminating critical body sounds. 2.3 History of Stethoscopes

The development of the stethoscope can be traced back to the beginning of the nineteenth century when a French physician by the name of Rene Laennec first invented the stethoscope in 1816. During Laennec's time, auscultation when undertaken, placed the physician's ear directly upon the patient's chest.4 For the sake of convenience and propriety, Dr. Rene Laennec simply rolled three pieces of paper and created a cylinder, placing one end on the patient's chest and keeping the other end over his ear. This primitive stethoscope that he had invented would become "the first noninvasive diagnostic instrument in medical history" . The invention of the stethoscope resulted in, without precedent, the most widely spread diagnostic instrument in the history of biomedical engineering. The stethoscope has evolved over the years, but the underlying technology remains the same.

Throughout the twentieth century the stethoscope became an integral part of medicine. The beginning of this century was marked by the introduction of the electronic stethoscope which could be used to make records of heart sounds and murmurs and also for teaching purposes . There continued to be modifications of pre-existing designs as well as introductions of new designs, some of which were considered impractical while others proving to be useful. However, the invention of the binaural stethoscope in the 1850's has left its mark in history, as this ingenious invention is more or less the modern day form that is used today [6]. Technology developments show no sign of ending. Current and recent developments of the stethoscope are focused on obtaining the best acoustic properties of this binaural form. A selection of stethoscopes from different eras is shown in Figure 2.2.

2.4 Types of Stethoscopes

There are several basic types of medical diagnostic equipment. The stethoscope is most often used to listen to heart sounds and breathing. There are two basic types of stethoscopes for respiration system diagnostics of the human body.

2.4.1 Acoustic Stethoscope

Acoustic stethoscopes are familiar to most people, and operate on the transmission of sound captured by a chest piece, via two air-filled hollow tubes, to the listener's ears [8]. The chest piece usually consists of a bell (hollow cup) and a diaphragm (plastic disc) that can be placed against the patient for sensing sound. The bell is used with light skin contact to hear low frequency sounds and the diaphragm is used with firm skin contact to hear high frequency sounds. The disadvantage is that the sound level is very low, making these stethoscopes less practical in noisy environments. However, acoustic stethoscopes are the most commonly used.

2.4.2 Electronic Stethoscope

Electronic stethoscopes function in a similar way as acoustic stethoscopes, but the sound is converted to electrical signals which can then be amplified and processed for optimal listening.8 Because the sounds are transmitted electronically, an electronic stethoscope can be a wireless device, can be a recording device, and can provide noise reduction, signal enhancement, and both visual and audio output.

2.5 Physiology of the Heart

The human heart is composed of four chambers, see Figure 2.3. The upper chambers are called atria and the lower chambers are called ventricles. The heart muscle squeezes blood from chamber to chamber. At each squeeze, the heart valves open when each chamber contracts to let blood through to the next chamber and close to prevent backflow of blood when the contraction is completed [15]. In this way, the valves keep blood moving as efficiently as possible through the heart and out to the body.

2.6 Heart Sounds

The sound heard during auscultation are called the first (S1) and the second (S2) heart sounds respectively, shown in Figure 2.4. The first 2 heart sounds (S1 and S2) are “normal” heart sounds that should be detectable in most patients. The S1 sound represents the near-simultaneous closure of the mitral and tricuspid valves, after blood has returned from the body and lungs [9]. This is the start of systole. The S2 sound represents the near-simultaneous closure of the aortic and pulmonary valves as blood exits the heart to the body and lungs [9]. This is the end of systole and the beginning of diastole.

2.7 Heart Murmurs

A heart murmur is an abnormal sound of the heart that is usually caused by valvular dysfunctions which occurs when a valve does not work the way it should.10 Murmurs can occur in either systole or iastole, as seen in Figure 2.5. Systolic murmurs occur between the first and second heart sound (S1 and S2), and diastolic murmurs occur between the second and first heart sounds (S2 and S1). The everity of systolic murmurs 10 is determined by grades, with grade 1 being the lowest in amplitude and 6 being the highest, see table 2.1. Murmurs with higher amplitudes can sometimes be heard without a stethoscope. Most murmurs are not serious, and many childhood murmurs disappear with time.

2.8 Acquisition of Heart Sound Signal

The audio recording involves a sequence of transformations of the signal: a sensor to convert sound into electrical signal, a preamplifier to amplify the signal, a low pass filter to remove background noise, a power amplifier to amplify the filtered signal and an analogue to digital converter to convert the signal into digital form.

2.8.1 Sensors

There are multiple types of sensors that can be used in the chest piece of an electronic stethoscope to convert body sounds into an electronic signal. Microphones and accelerometers are the common choice of sensor for sound recording. These sensors have a high-frequency response that is quite adequate for body sounds. Rather, it is the low-frequency region that might cause problems [11]. The microphone is an air coupled sensor that measure pressure waves induced by chest-wall movements while accelerometers are contact sensors which directly measures chest-wall movements [12]. For recording of body sounds, both kinds can be used. More precisely, condenser microphones and piezoelectric accelerometers have been recommended, see Figure 2.6 [13].

Both transducers are popular in sound recording. However, accelerometers are typically more expensive than microphones, are often fragile, and may exhibit internal resonances. Thus, this concludes that the microphone is prefect for the application.

2.8.2 Filters

The sensors used in electronic stethoscopes often pick up background noise, therefore it is important to filter that signal and only reproduce the sounds of interest. There are two kinds of filter, analog and digital, that can be used in an electronic stethoscope to reduce the ambient noise and increase the intensity of the desired signal. Analogue Filter

An analog filter uses analog electronic circuits made up from components such as op-amps, capacitors, and resistors to produce the required filtering effect. The disadvantage of the analog filter is that large number of components may be needed to implement a filter that has sufficient roll-off at the desired cutoff frequency. Also, the multiple analog components required can add noise to the signal. Digital Filter

Digital filters, however, can be implemented without added noise. The two digital filter types, finite impulse response (FIR) and infinite impulse response (IIR) can be implemented using software. Digital filters with FIR filters have both advantages and disadvantages compared to IIR filters. FIR filters are always stable, while IIR filters may be unstable. However, the primary advantage of IIR filters over FIR filters is that they typically meet a given set of specifications with a much lower filter order than a corresponding FIR filter. Although IIR filters have nonlinear phase, data processing within Matlab software is commonly performed "offline," that is, the entire data sequence is available prior to filtering. This allows for a non-causal, zero-phase filtering approach, which eliminates the nonlinear phase distortion of an IIR filter. Flexibility

Analogue filters commonly require more effort to change than digital filters. For example, changing the cutoff frequency of an analog filter will often require a component change. A digital filter is extremely flexible, allowing changes to any aspect of the filter. However, this does not mean that analogue filtering will not be used in the stethoscope.

2.9 Analysis of Heart Sound Signal 2.9.1 Time and Frequency Analysis

The traditional method to analyze the signal is to view them in time-domain representation by showing parameter variation such as amplitude versus time. The other representation is the frequency analysis; also know as spectral analysis, which shows the frequencies existed for total duration of time. There are several researchers using time and frequency analysis to examine the heart sounds.

In general, time and frequency analysis provide fundamental information to the signal analysis. However, time and frequency analysis do not fully describe the whole interpretation of the signals of interest such as heart sounds. For instance, the classical spectral analysis needs to be enhanced to analyze the signals. Thus, another type of representation should be considered for signal analysis such as time-frequency analysis.

2.9.2 Time- Frequency Analysis

Time-frequency analysis proves to be a powerful method due to its ability to determine which frequencies existed at a particular of time compared to frequency analysis that can only show the range of frequency component of the signals. The time or frequency analysis alone is unable to represent the multi-component signals in logical way. Time-frequency analysis reveals the multi-component nature of the signals by concentrating the signal energy in time-frequency plane around the instantaneous frequency law component.



This stage is critical to successful resourcing and execution of the project activities and it includes the development of the overall project structure, the activities and workplan/timeline that will form the basis of the project management process throughout the project lifecycle.

The electronic stethoscope must be carefully evaluated and implemented. The hardware and software that are most cost effective must be identified and implemented. Through a careful and thorough planning, an electronic stethoscope can be designed and constructed in time.

An action plan and milestones must be prepared. These identify the many actions required to implement an electronic stethoscope. Although the schedule is important, several other elements should be addressed, including the types of resources required to complete the action, funding sources for the implementation, and the access to equipment and facilities. The milestones identified for the project are the significant steps along the path to implementation.

3.1 Scope of Project

The primary focus of this project is to develop and construct an electronic stethoscope that will make it easier to detect heart sounds. Standard stethoscopes provide no amplification which limits their use. This stethoscope will be designed with a certain level of amplification and includes a low pass filter to remove ambient noise.

Signals from stethoscope will be transferred to the microphone, filtered and amplified. Then signals will be fed into an A/D converter and convert them into digital signals to be displayed on the computer using Matlab program for analysis.

3.1.1 Project Tasks

The table below shows the list of key tasks required in this project.

Task 1

Background study and literature review on stethoscope

Task 2

Design a power circuit

Task 3

Design a preamplifier circuit

Task 4

Design a low-pass filter

Task 5

Design a power amplifier circuit

Task 6

Overall schematic generation

Task 7

Implementation of the circuit

Task 8

Writing the software algorithm

Task 9

Testing and Debugging

Table 3.1: Project Tasks List

3.1.2 Project Management Plan

A good project management is one of the keys to a successful project outcome. The project work was distributed in the Gantt chart shown below.



The hardware for the stethoscope composes of 2 parts: sensor and signal conditioner. The signal conditioner is further divided into 3 stages: preamplifier, low-pass filter and power amplifier.

4.1 Architecture of the Stethoscope System

An overview of the stethoscope is shown below.

4.2 Hardware Design

The hardware design of this project is composed of the following major parts: power supply, sensor, preamplifier, low-pass filter and power amplifier. In the sections that follow, the specifications of each part will be described.

4.2.1 Power Supply

Most portable systems have one battery, thus, the popularity of portable equipment results in increased single supply applications. Although it is advantageous to implement op amp circuits with balanced dual supplies, there are many practical applications where, for energy conservation or other reasons, single-supply operation is necessary or desirable. But single-supply operation has its drawbacks: It requires additional passive components in each stage and, improper execution of the design can lead to serious instability problems.

The circuit in Figure 4.2 shows a single-supply biasing method. 9V is chosen since it is compatible with the microphone, op amp and power amplifier. This non-inverting, op‑ amp circuit uses a resistor ivider with two biasing resistors, R5 and R6, to set the voltage on the non-inverting input equal to Vcc/2. R5 and R6 are equal values, selected with power consumption versus allowable noise in mind

The values of R5 and R6 are chosen to be as low as feasible; the 47 kΩ values chosen here are intended to conserve supply current. Attempts to use small resistor values in the voltage divider will increase power-supply current consumption, may overheat the resistors, and certainly is not a good design approach.

To avoid substantial feedback through the power supply at low signal frequencies, a larger capacitor is needed to effectively bypass the voltage divider at all frequencies within the circuit's passband. Therefore the capacitance value of 1μF for C3 is chosen.

With a 47kΩ/47kΩ voltage divider for R5 and R6 and a 1μF capacitance value for C3, the -3dB bandwidth of this network's impedance, set by the parallel combination of R5,

1 1

R6, and C3, is equal to = H

7 z

( RA (

47 k


3 \

1 2 )1 1 2 ,r1 1ts

2 2

The common-mode rejection drops below 7Hz. Instability will not occur as the electret condenser microphone only picks up signal above 20Hz.

In figure 4.3, it shows a noisy voltage with a bypass capacitor installed. Random electrical noise causes the voltage to fluctuate. This is often called 'noise' or 'ripple'. The blue line, represents the voltage of a circuit that doesn't have a bypass. The pink line is a circuit that has a bypass. Ripple voltages are present in almost any DC circuit. Even with the bypass, the voltage does fluctuate, even though it is to a smaller degree. The key function of the bypass capacitor is to reduce the amount of ripple in a circuit. Too much ripple is bad, and can lead to failure of the circuit. Ripple is often random, but sometimes other components in the circuit can cause this noise to occur. By installing bypass capacitors, the DC circuit will not be as susceptible to ripple currents and voltages.

C12 is a bypass capacitor which on top of preventing the supply voltage from fluctuating, also supplies positive supply current when the battery runs down and its internal resistance rises.

4.2.2 Sensor

The quality of an acoustic stethoscope is greatly affected by the quality of its chest piece. Due to the lack of expertise necessary to design the mechanical aspects of a chest piece, modification is done on an existing chest piece. The microphone is placed within it to maximize the benefits of mechanical filtering that are already present.

As mentioned in Part 2, microphone was chosen for this project. The next step is to choose which specific microphone to use. In this decision there are several criteria that must be examined. One of the criteria is the microphone's frequency response. It must be able to pick up sounds in the range of 20Hz to 2kHz. Another criterion is its sensitivity;

the microphone must be able to reproduce the correct intensity of body sounds that it receives.

Dynamic Microphone

Condenser Microphone

Do not have flat frequency response

Have a flat frequency response

Operate with the principle of
Electromagnetism as it does not require
voltage supply.

Employs the principle of electrostatics
and consequently, require voltage supply
across the capacitor for it to work.

It is suitable for handling high volume
level, such as musical instruments.

It is not ideal for high volume work as its
sensitivity makes it prone to distortion.

The signal produced are strong therefore
making them sensitive

The resulting audio signal is stronger
than that from a dynamic. It also tends to
be more sensitive and responsive than

Table 4.1 Comparison between Dynamic And Condenser Microphone

Table 4.1 shows the comparison between the dynamic and condenser microphone. Condenser microphones generally have flatter frequency responses than dynamic, and therefore mean that a condenser microphone is more desirable if accurate sound is a prime consideration as required in this design.

There are two types of condenser microphones; standard condenser and electret condenser. A standard condenser microphone consists of a small diaphragm that vibrates in response to acoustic pressure. Standard condenser microphones have very high output impedance, so they are not suitable for transferring signals over even a very small distance. An electret condenser microphone combines a condenser microphone with a Field Effect Transistor (FET), which amplifies the signal and transforms the impedance

to a more useful level. This characteristic of electret condenser microphones makes them very sensitive to small sounds. Its properties can be found in Appendix C.

The electret condenser microphone requires a supply voltage in order to bias the FET inside the microphone. Most electret condenser microphones have only two prongs, so the common way of providing the bias current to the microphone is known as Phantom Power. The basic procedure is to use a resistor between the voltage source and ground to limit the current into the FET, and a capacitor to block the DC offset of the supply voltage from the amplifier circuitry.

R1 is to provide bias current for the electret microphones internal FET The bias voltage of around 1-10V is needed to supply the build-in FET buffer. The load resistor defines the impedance and can be matched to the low noise amplifier intended. Therefore, the suitable resistance values are typically in the range of 1-10 kΩ. The lower limit is defined by amplifier voltage noise and the upper limit by interference pickup (and amplifier current noise). R2 and C1 is a supply filter circuit to keep supply voltage fluctuations away from the circuit's sensitive input.

4.2.3 Preamplifier

An operational amplifier, which is often called an op-amp, is a DC-coupled high-gain electronic voltage amplifier with differential inputs and, usually, a single output. Typically the output of the op-amp is controlled either by negative feedback, which largely determines the magnitude of its output voltage gain, or by positive feedback, which facilitates regenerative gain and oscillation. High input impedance at the input terminals (ideally infinite) and low output impedance (ideally zero) are important typical characteristics. The op-amp exhibits high input impedance and thus when placed in parallel with the body's relatively low impedance is assumed to allow the following section to amplify the voltage with minimal current consumption.

A couple of different operational amplifiers have been specified for the preamplifier. Table 4.2 shows a comparison of the suggested op-amps.









±6V - ±22V







±2.5V - ±18V







±7V - ±18V






Table 4.2: Comparison of Operational Amplifiers

The OPA2134 has the most excellent performance among the 3 op-amps. However, it is naturally more expensive and is not available in local shops. Therefore, NE5532 is chosen because of its availability and has lower noise as compared to TL072. However, it is power hungry since it has high input bias currents.

The NE5532 is an internally compensated dual low noise OP-AMP. The high small signal and power bandwidth provides superior performance. It is also a low-power device that can be operated from a single voltage supply, therefore appropriate for battery-operated circuits.

The gain is only a function of the feedback and gain resistors, so the feedback has accomplished its function of making the gain independent of the op amp parameters.

The impedance level does not set the gain; the ratio of R4/R3 does. The circuit input impedance is set by R3 because the inverting input is held at a virtual ground.

4.2.4 Low Pass Filter

In this project, it is desired to filter out the high-frequency sounds picked up by the
microphone which make it difficult to listen to (low-frequency) bodily sounds (e.g., a
heart beating). As such, the frequency response specifications do not need to be

extremely accurate since we are dealing with audible frequencies and the human ear cannot discern frequencies that are close together [16]. Filters that use op-amps as the active element provide several advantages over passive filters (R, L, and C elements only). The op-amp provides gain, so that the signal is not attenuated as it passes through the filter. The high input impedance of the op-amp prevents excessive loading of the driving source, and the low output impedance of the op-amp prevents the filter from being affected by the load that it is driving. Active filters are also easy to adjust over a wide frequency range without altering the desired response.

The Sallen-Key filter is used since it is a popular filter due to its versatility and ease of design. The Sallen-Key is one of the most common configurations for a second-order (two-pole) filter.

1 1

Letting R 1 = R 2 = R , and C 1 = C 2 = C , results in: fc = and Q = . Now



and Q are independent of one another, and design is greatly simplified although limited. The gain of the circuit now determines Q . RC sets - the capacitor chosen and the


resistor calculated. One minor drawback is that since the gain controls the Q of the circuit, further gain or attenuation may be necessary to achieve the desired signal gain in the pass band.

In Figure 4.6, it shows a precision 2-pole Sallen-Key Butterworth low-pass filter that cuts-off frequencies above 100Hz and 1000Hz. Heartbeat and respiration sounds are passed and background sounds are reduced. A frequency switch for heart sounds (20~100Hz) and respiration sounds (20~1000Hz) will be incorporated which allows the listener to concentrate on a particular sound.

For heart sounds (20~100Hz),

Since R 1 = R2 = 33k  andC 1 = C2 = 47nF,

For respiration sounds (20~1000Hz),

Since R 1 = R2 = 33k  and C 1 = C2 = 4. 7 nF,

The op-amp in the second-order Sallen-Key filter acts as a non-inverting amplifier with the negative feedback provided by the R9/R10 network. The damping factor is set by the values of R9 and R10, thus making the filter response Butterworth, Chebyshev, or Bessel. Table 4.3 shows a comparison of the filter responses. The R9/R10 ratio must be 0.586 to produce the damping factor of 1.414 required for a second-order Butterworth response.

For a Butterworth response,

R 9 / R10 = 0. 5 8 6

By choosing R10 = 56kΩ,

R 9 0 . 5 86( 1 0) 0 . 5 86(5 6 ) 33

= R = k 1 k 1

R 9

The overall gain is set by the ratio of R9 and R10. The gain is 1 . 6


R 10



Butterworth Response

It provides maximally flat
magnitude response in the
pass-band. It has good all‑
around performance. Its
pulse response is better than
Chebyshev. Its rate of
attenuation is better than
that of Bessel.

Some overshoot and ringing
is exhibited in step

Chebyshev Response

It provides better
attenuation beyond the
pass-band than Butterworth.

Ripple in pass-band may be
objectionable. There is
considerable ringing in step

Bessel Response

It provides best step
response: very little
overshoot or ringing.

It exhibits slower rate of
attenuation beyond the
pass-band than Butterworth.

Table 4.3: Comparison of Various Filter Responses

4.2.5 Power Amplifier

There are two primary factors that control the frequency response of an amplifier circuit; these are the choice of active components (such as transistors and ICs), and the design of input and output coupling networks.

The LM386 circuit is an audio amplifier designed for use in low voltage consumer applications which provides both voltage and current gain for signals. The inputs are ground referenced while the output automatically biases to one-half the supply voltage. The quiescent power drain is only 24 milliwatts when operating from a 6 volt supply, making the LM386 ideal for battery operation.

This chip has been a popular choice for low-power audio applications for many years. There are many other audio amp ICs on the market, but the LM386 is sufficient for the purposes. Another benefit about the LM386 is that the gain-frequency curve can be shaped with some external feedback components, so it is a very flexible device.

The gain is internally set to 20 to keep external part count low, but the addition of an external resistor and capacitor between pins 1 and 8 can increase the gain to any value from 20 to 200.

All the capacitors and resistors were non-critical except for the 1000uF output coupling capacitor, C11, which couples the audio but blocks the half-supply DC voltage at the output of the LM386 from going to the earphones. The other values are only required for the Zobel network which neutralizes the effect of the voice coil's inductance. The diodes, D1 and D2 clamp the input voltage on pin 3 at +/- 0.7V, to ensure that excessive voltage is never applied to this pin, which could damage the circuit. The 1K value for R13 fully-charges C11 in 1 second.

4.2.6 Combination of Individual Circuits

Combining all the individual parts to complete the circuit often leads to unavoidable interactions between filter response characteristics, noise, and other circuit characteristics. It should always begin by prototyping separate gain, offset, and filter stages, and then combine them if possible after each individual circuit function has been verified. The overall schematic of the circuit can be found in Appendix A.

Gain from preamplifier circuit = 21

Gain from Sally-Key low-pass filter = 1.6

Gain from power amplifier = 20

21 1 . 6 20

x x

Overall gain = 336


The reason why the overall gain was divided by half was due to the voltage divider that cut the gain to half.

4.3 Software Design

The software for this project is completely designed in Matlab.

4.3.1 Graphical User Interface

In order to meet the objectives stated in Chapter 1, a Graphical User Interface (GUI) was developed. Figure 4.8 shows the user interface.

The full program codes can be summarized into the following main modules;

  1. Recording of heart beat and respiration signals from the hardware prototype
  2. Displaying the signals in different waveforms
  3. Displaying of heart beat rate
  4. Stores the data in the computer
  5. Replaying of the data stored
  6. Filtering the signal
  7. Enable user to choose the recording quality

This GUI is divided into one main GUI and two sub GUIs. The main GUI will include all of the above main modules except 6 and 7. One of the sub GUIs is to allow the user to choose the recording quality. The other sub GUI allows the user to choose the filtering parameters. The Matlab code can be found in Appendix D to F.

4.3.2 Time Analysis

A signal is generally a function of many variables. Signal in a function of time is the most fundamental analysis as it displays the signal amplitude varying in time. Time-domain signal allows other properties to be determined such as energy and power of the signal. Figure 4.9 shows a normal heart sound in time-domain representation.

4.3.3 Frequency Analysis

Spectrum analysis is one of the basic methods to analyze signal that is also referred to as the frequency analysis. The signal is decomposed and represented in terms of its frequency components. The spectrum analysis of heart sound is shown in Figure 4.10.

Based on Figure 4.10, the power spectrum of heart sound is able to show frequency content of the signal. However, this method is unable to localize the frequency content prior to time-varying characteristic in the time-domain representation and failed to determine the frequency at a particular time. Thus, an alternative method will be required to display and differentiate various types of heart sounds.

4.3.4 Time-Frequency Analysis

Time-frequency analysis is chosen to analyze time-varying and non-stationary signals such as heart sounds. It has been used successfully in applications such as speech analysis, communication and biomedicine signals. Spectrogram

One of the best-known time-frequency distributions is the spectrogram, defined as the squared magnitude of the short-time Fourier transform. The spectrogram of heart sound is shown in Figure 4.11.

4.3.5 Digital Filter

A low-pass digital filter is designed using various analog prototypes: Butterworth, Chebyshev and Elliptic. The optimum filter type is chosen on the basis of implementation complexity, magnitude response and phase response. In terms of passband ripple, the Butterworth filter gives the optimum response. The frequency response of the Butterworth filter is maximally flat (has no ripples) in the passband, and rolls off towards zero in the stopband. The elliptic and Chebyshev filters both have much more ripple in the passband. So, there is a tradeoff between these three different types of filters. Butterworth filter is most likely the optimum choice since it have a more linear phase response in the passband than the Chebyshev and elliptic filters. Figure 4.12 shows one of the sub GUIs, which allows the user to choose the parameters for filtering.

4.3.6 Heart Rate Calculation

Heart rate is calculated using simple threshold and peak finding. The flow chart of the heart rate calculation can be found in Appendix B.

4.3.6 Quality of Audio Recordings

The dynamic range of the heart and lungs is 20Hz to 1kHz. In order to preserve sound quality during recording, sampling should be done at a rate that is at least twice the highest frequency of interest (1kHz in the case of the heart and lungs). This is known as the Nyquist sampling rate, the rate at which no aliasing occurs. Of course it is better to sample at a rate higher than 2kHz, for improved sound quality. Figure 4.13 shows the user interface of the recording option. The recording option is one of the sub GUIs, which is called from the main GUI in Figure 4.8. It allows the user to change the sampling frequency and bit resolution.



The hardware design which composes of the following major parts: power supply, sensor, preamplifier, low-pass filter and power amplifier were individually simulated and/or tested before the hardware prototype was completely assembled as shown in Figure 5.1.

The overall circuit was first constructed on a breadboard, shown in Figure 5.2, to debug and demonstrate that it works properly. The idea of using a breadboard to test the circuit first is a good practice in general, since it allows debugging of the circuit design more easily and allows to experiment with different component values.

5.1 Performance of Power Supply

As one can see, there is a lot of high frequency noise displacing the DC level (approximately 10mVP-P). Then, far more pronounced, there are regular spikes in excess of 50mV. Since power supplies are assumed to be stable, any perturbations will couple directly into the circuit and might cause instability.

The effect of the bypass capacitor on the stability of the output of the non-inverting amplifier can be seen in Figure 5.2.

5.2 Performance of Low-Pass Filter

5.3 Results of Heart Beat Calculation

The recordings were made in a normal room (some background noise is present in some recordings). The patients were instructed to breathe normally. Each recording was 10 seconds long, recorded in 16 bits resolution and sampled with 8000 samples per second. Recordings from 5 subjects were included. The recordings were low-pass filtered with a 2nd order Butterworth filter using a cut-off frequency at 200Hz. The records are shown in Table 5.1.

Measured Heart
Beat Rate

Actual Heart Beat

Accuracy (%)

Patient 1




Patient 2




Patient 3




Patient 4




Patient 5




Table 5.1: Accuracy of the Heart Beat Calculation

5.4 Digital Filtering of Heart Signals

5.5 Problems Encountered

Below is a list of problems encountered during the development of the hardware prototype.

1Earthing Problem

During the testing of the hardware prototype, there was a loud buzz coming out from the output. The buzz is present with or without an input signal. Voltage check was done across the circuit and found that it was not giving any problem. The op-amp and power amplifier were replaced but to no available. Finally, the soldering of the components was

checked and found that the log potentiometer was not grounded properly.

2Bad Layout and Soldering of Circuit

After building the hardware prototype on an actual PCB, no output signal was detected. Voltage check across the circuit was done and found that the op-amp was not powered correctly and a few components were not grounded properly. The output signal was detected after the problems were fixed. However, the signal had a lot of noise. Therefore, another prototype was remade as shown in Figure 5.1.

5.6 Shortcoming of the Project

Owning to time constraints and partly the lack of maintaining self discipline and commitment, some of the individual circuit parts were not tested. More details will be defined in the next chapter under critical review and reflection.



6.1 Conclusion

Single-supply op amp design is more complicated than split-supply op amp design, but with a logical design approach excellent results are achieved. Single-supply design was considered technically limiting because the older op amps had limited capability. Single-supply op amp design usually involves some form of biasing, and this requires more thought. Therefore, designing a single supply op amp circuit design that considers input bias current errors as well as power supply rejection, gain, input and output circuit bandwidth, etc., can become quite involved. So, single-supply op amp design needs discipline and a procedure. More attention must be paid to the details to achieve a successful single supply design.

The signal-to-noise ratio has several components that have to be analyzed. The signal comes to the op amp with a noise burden caused by the transducer, cabling, and connections. Making the op amp a filter/amplifier combination eliminates some of this noise. There is always system noise, and a portion of this noise propagates through the op amp into the signal. The system noise is minimized by extensive use of decoupling capacitors.

The heart beat rate calculation using simple threshold and peak finding is not as accurate as expected.

Overall, this project was a really enriching experience. The process of this project has
certainly equipped me with the necessary skills by research information, selecting

components, design circuit, calculation, testing and troubleshooting of the circuits. The issues that arise had certainly provided me with the platform for a problem based learning experiences.

Cardiac auscultation remains an important diagnostic tool, when performed skillfully, can provide clinicians with a wealth of information regarding patients' cardiac health. However, the standard acoustic stethoscope which has been useful for more than a century, cannot process, store, and play back sounds or provide visual display, and teaching is hindered because there is no means to distribute the same sounds simultaneously to more than one listener. Modern portable and inexpensive tools are now available to provide, through digital electronic means, better sound quality with visual display and the ability to replay sounds of interest at either full or half speed with no loss of frequency representation or sound quality. Visual display is possible in both standard waveform and spectral formats. Heart sounds are easily archived and retrieved, in addition, would allow physicians to keep an archive of retrievable auscultatory examinations that would allow physicians to track the progression of cardiovascular disease in patients based on auscultatory findings. Permanent auscultatory records could be included as part of a patients medical record and may be used to support referrals for more costly and specialized diagnostic procedures.

6.2 Future Work

No project is ever complete for there is always room for improvement. Throughout the different stages of the progress, some of the areas that could benefit from improvements were noted down.

The obvious deficiencies with this project are:

1) It is monophonic (not stereo): For a true stereo output, a separate amplifier is needed

2) It is not very efficient in terms of battery life: It would be better to use OPA2134 as it consumes a lot less power than NE5532.

3) Low roll-off rate from the low-pass filter: To achieve higher roll-off rate, it can be done by cascading two two-pole low-pass filters as shown in Figure 6.1. The reason for only using a 2 pole filter is also to reduce the size of the circuit board.

4) Heart beat rate calculation not near prefect: By using a more complex algorithm such as autocorrelation, the heart beat rate calculation can be more accurate.

Advances in technology with the introduction of electronic stethoscopes mean that early detection of heart disease is fast becoming a reality in primary care settings. Evaluating the acoustic properties of the heart using auscultation methods with a traditional stethoscope has long been the trusted method of assessment. However, the continually evolving technology over the past ten years of microchip development means that electronic stethoscopes have the facility to analyze and measure heart sounds in a more objective manner. Now practitioners can rely less on their subjective judgement and own hearing ability to gain an accurate representation of sounds using the electronic version.

Software versions allow downloads to computer after digitally recording heart sounds for data analysis. Noise reduction technology reduces noise by 75% on average but still allows critical body sounds to be heard. Researchers now use this facility to provide valuable new insight into the diagnostic value of the heart sounds and the analysis techniques adopted using PC downloaded recordings include wavelet transform and neural network. It is anticipated that with further technical advances in the field of the development of electronic models, the possible detection of heart disease early in a patient's examination is likely to be a scenario seen far more predominantly in a physician's office.

6.3 Critical Review and Reflections

To review upon the project and how it has progressed, I have to summarize by admitting the level of design, prototyping and user involvement was considered low. I would admit that the concept has gradually become more and more dilute due to time constraints and lack of self discipline and commitment. However, I do feel that this project is useful, and would be extremely well picked up on if it was applied correctly.

The skills which I need improvement were project planning, understanding of Matlab and PSpice, and report writing. In order to tackle these weaknesses, I did some read up online. Even though I did some reading, but still, I failed to tackle weaknesses in report writing and project planning. In general, I am not very pleased with the lack of user involvement and prototype testing.

To reflect on myself briefly, I have not applied myself or committed myself 100% to this project, mainly due to other commitments. I feel I am not committed enough to stick to my Gantt chart and follow the schedule; therefore, there are lapses in my work.

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