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Introduction The operational amplifier has a non-inverting input terminal and an inverting input terminal in the electronic circuit. The polarity of the input terminal and the output terminal are the same is a non-inverting amplifier, and the polarity of the input terminal and the output terminal are opposite called inverting amplifier. The inverting amplifier circuit has the function of amplifying the input signal as inverting output. Inverting Op Amp and The Concept of Virtual Ground in Op Amp Catalog Introduction Ⅰ Inverting Amplifier Basics Overview 1.1 Working Principle 1.2 Inverting Amplifier Gain Calculator 1.3 Inverting Amplifier Features 1.4 Inverting Amplifier Functions Ⅱ Inverting Amplifier Applications Ⅲ Inverting Amplifier Explain with Diagrams: NE5532 Ⅳ Inverting Amplifier Circuit Design Steps Ⅴ FAQ Ⅰ Inverting Amplifier Basics Overview 1.1 Working Principle As shown in Figure 1, the inverting amplifier circuit has the function of amplifying the input signal as inverting output. "Inverting" means that the positive and negative signs are reversed. This amplifier uses negative feedback technology, which used to return a part of the output signal to the input. In Figure, the wiring method of connecting the output Vout to the inverting input terminal (-) via R2 is negative feedback. Figure 1. Inverting Amplifier Circuit Operational amplifiers have such characteristics. When the power supply voltage is not applied to the output terminal, the non-inverting input terminal (+) and the inverting input terminal (-) are considered to have the same voltage, that is to say, it can be regarded as a virtual short circuit. Therefore, when the positive input terminal (+) is 0V, the voltage at output is also 0V.The input impedance of the operational amplifier is extremely high, and there is basically no current in the inverting input terminal (-). Therefore, when the current flows to R2, the I1 and I2 are basically equal. Based on the above conditions, using Ohm's Law for R2, we get Vout=-I1xR2. I1 is negative because I2 flows from point A where the voltage is 0V. From another point of view, when the input voltage of the inverting input terminal (-) rises, the output will be inverted and amplified greatly in the negative direction. Since the output voltage in the negative direction is connected to the inverting input terminal via R2, the voltage rise of the inverting input terminal (-) will be blocked. Both the inverting input terminal and the non-inverting input terminal voltage become 0V, and the output voltage is stable.   1.2 Inverting Amplifier Gain Calculator Calculate the gain through the relationship between the input and output in this amplifier circuit. The gain of an inverting op amp is the ratio of the feedback resistance to the input resistance, that is, the ratio of Vout to Vin, and the formula is Vout/Vin= (-I1xR2) /(I1xR1)=-R2/R1. The resulting gain is negative, indicating that the waveform is inverting.The current flowing through R1: I1=(Vi-V-)/R1...aThe current flowing through R2: I2=(V--Vout)/R2……bV-=V+=0………………cI1=I2……………………dSolve the above algebraic equations to get Vout=(-R2/R1)*Vi.This is the input and output relationship of the inverting amplifier circuit.   1.3 Inverting Amplifier Features It can reduce the input impedance or keep a certain value.It can be used as a current input type.Virtual short-circuit point is generated at a certain potential.Its positive input port is free.If the signal source impedance is low, it is easier to obtain a required S/N.The magnification is -Rf/R.   1.4 Inverting Amplifier Functions The inverting amplifier is the basic gain stage in the CMOS circuit. It adopts a common source structure, and the load can be an active load or a current source.Advantages: The potential of the two input terminals is always approximately zero (the non-inverting terminal is grounded, and the inverting terminal is virtual ground). With only differential mode signals,it has strong anti-interference ability.Disadvantages: The input impedance is very small, equal to the resistance of the series resistance from the signal to the input.   Ⅱ Inverting Amplifier Applications Figure 2. Basic Inverting Amplifier Circuit 1) As a IntegratorThe original resistor R2 of the inverting amplifier is replaced by a capacitor C2. At this time, the relationship between the input signal Vi and the output signal Vo forms an integral relationship.2) As a DifferentiatorReplace the original resistor R1 of the inverting amplifier with an electric capacitor C. At this time, the relationship between the input signal Vi and the output signal Vo is differential.3) As a AdderIf the inverting amplifier is slightly changed, the relationship between the input signal and the output signal Vo at this time, if R1 = R2 = R3 =...= Rn = Rf, it can be simplified to Vo = -(V1+V2+V3+.. .+Vn), is additive.   Ⅲ Inverting Amplifier Explain with Diagrams: NE5532 The equivalent resistance seen between the input terminal and the ground of the inverting amplifier circuit is equal to the equivalent resistance between the input terminal and the virtual ground, so the input resistance of the circuit is Ri=R. It can be seen that although the input resistance of the ideal operational amplifier is infinite, the input resistance of the inverting proportional arithmetic circuit is not large because of the parallel negative feedback introduced by the circuit. Figure 3. NE5532 Audio Amplfier Circuit In order to increase the input resistance, R must be increased. For example, when the scale factor is -50, if Ri=10kΩ, R should be 10kΩ and Rf should be 500kΩ. If Ri=100kΩ, R should be 100kΩ and Rf should be 5MΩ. In fact, when the resistance in the circuit is too large, on the one hand, due to the process, the stability of the resistance is poor and the noise is large, on the other hand, when the resistance is of the same order of magnitude as the input resistance of the integrated op amp, the proportional coefficient -Rf/R of the circuit will change greatly, and its value will not only be determined by the feedback network. Therefore, it is necessary for practical applications to use a resistor with a smaller resistance value to get a larger scale factor and a larger input resistance.Look at the following two simulations: Figure 4. Current and Voltage Measuring From above, it doesn't matter without R0, take a look at this below. Figure 5. Current and Voltage Measuring Their output is different, if the expected result is -4V. The reading value -3.608V obviously does not meet the design requirements. We now know that the resistance value of Rs in Figure 5 is orders of magnitude greater than the input resistance of the integrated operational amplifier. Next, if have R0, look at the picture below. Figure 6. Current and Voltage Measuring It can be seen from Figure 6 that after adding R0, the output of the op amp is normal. Why is this?In a TTL circuit, the transistor has a bias current, which will produce a DC voltage drop on the feedback resistor of the e pole. Although it is small, the amplifier has a high amplification factor, which affects the output accuracy greatly. A resistor in the same value as the negative terminal is connected to the positive terminal to cancel the effect of the bias current.R0 is a compensation resistor, which minimizes the bias current error to ensure the symmetry of the differential amplifier circuit of the input stage. When its value is u=0 (that is, the input terminal is grounded), the inverting input terminal is always the equivalent resistance, that is, the parallel connection of the resistance of each branch, so R0=Rs/Rf, through Rf in the circuit, induce negative feedback. According to practical experience, it is best to add a compensation resistor to improve the stability of the circuit. Of course, it can be omitted under normal circumstances.   Ⅳ Inverting Amplifier Circuit Design Steps Inverting amplifiers are several commonly used amplifier types. How to use operational amplifiers design inverting amplifier circuit? The following sharing is the detailed steps.Step 1: Determine the magnification.As shown in the figure below, the amplification factor of the inverting amplifier is (-R2/R1). Figure 7. Determine the Magnification Step 2: Determine the supply voltage.Ensure that the output voltage after the input voltage Vin is multiplied by the amplification factor (-R2/R1) will not exceed the power supply voltage. If it is not the rail-to-rail operational amplifier used, it is better to have a margin of 1~2V. Figure 8. Determine the Supply Voltage Step 3: Determine the gain bandwidth product (GBW).If you are amplifying an AC signal, you need to consider the gain bandwidth product. The calculation formula is: bandwidth of input signal × design gain, select an operational amplifier whose GBW is greater than the required one. Step 4: Check the bandwidth with the slew rate (SR).If the amplifying is a large AC signal, there may be insufficient bandwidth, and it may not be accurate enough to calculate based on the gain-bandwidth product.SR = 2*pi*f*Vp -> f = SR/(2*pi*Vp), where Vp: peak output voltage.If the calculated value f is less than the input voltage frequency, the operational amplifier is not suitable and needs to be replaced. The actual bandwidth should be the smaller of the gain bandwidth product and the slew rate. Step 5: Determine the input offset voltage.Since the offset voltage is also a DC signal, if it is to amplify the DC signal, it should be noted that the offset voltage will also be amplified by the corresponding multiple. If the accuracy is high, try to choose an operational amplifier with a small offset voltage. Step 6: Determine the resistance value.In step 1, the ratio of R1 and R2 has been determined. R1 is generally 1k~10k. A small value is prone to cause gain errors, and a larger value will increase noise (resistor thermal noise). R2 takes the resistance value corresponding to the multiple of R1.It is better to choose the chip resistor, because the parasitic parameters are small.R3=R1/R2, if R1 and R2 are connected in parallel, which are not required.   Ⅴ FAQ 1. What is the inverting amplifier?An inverting op amp is an operational amplifier circuit with an output voltage that changes in the opposite direction as the input voltage. In other words, it is out of phase by 180o。   2. What is inverting amplifier and its application?The inverting amplifier is an important circuit configuration using op-amps and it uses a negative feedback connection. An inverting amplifier, like the name suggests, inverts the input signal as wells as amplifies it.   3. How does an inverting amplifier circuit work?In an inverting amplifier circuit, the operational amplifier inverting input receives feedback from the output of the amplifier. Assuming the op-amp is ideal and applying the concept of virtual short at the input terminals of op-amp, the voltage at the inverting terminal is equal to non-inverting terminal.   4. What are the applications of inverting amplifier?op-amp inverting amplifier. Op amp summing amplifier: Based around the inverting amplifier circuit with its virtual earth summing point, this circuit is ideal for summing audio inputs. It is widely used in audio mixer and many other applications where voltages need to be summed.   5. Why is it called inverting amplifier?It is called Inverting Amplifier because the op-amp changes the phase angle of the output signal exactly 180 degrees out of phase with respect to input signal. Same as like before, we use two external resistors to create feedback circuit and make a closed loop circuit across the amplifier.   6. What are the characteristics of inverting amplifier?1) No Current Flows into the Input Terminals.2) The Differential Input Voltage is Zero as V1 = V2 = 0 (Virtual Earth)   7. What is the formula of inverting amplifier?One final point to note about the Inverting Amplifier configuration for an operational amplifier, if the two resistors are of equal value, Rin = Rƒ then the gain of the amplifier will be -1 producing a complementary form of the input voltage at its output as Vout = -Vin.   8. What are advantages and disadvantages of inverting amplifier?Advantages and Disadvantages of Inverting AmplifierIt follows the negative feedback. The gain factor of these amplifiers is very high. The output generated will be out of phase with the applied input signal. The potential values at both the inverting and the non-inverting terminals maintained at zero.   9. What are the advantages of inverting amplifier?The op amp circuit for the inverting amplifier offers many advantages including relatively low input impedance, a low output impedance and the level of gain that is required (within the limits of the op amp and the gain required from the overall circuit.   10. What is the use of inverting amplifier?The inverting amplifier is an important circuit configuration using op-amps and it uses a negative feedback connection. An inverting amplifier, like the name suggests, inverts the input signal as wells as amplifies it.   11. What is mean by inverting amplifier?An inverting amplifier takes an input signal and turns it upside down at the op amp output. When the value of the input signal is positive, the output of the inverting amplifier is negative, and vice versa. ... The amount of amplification depends on the ratio between the feedback and input resistor values.   12. How an opamp is used as inverting amplifier?Theory: An inverting amplifier using opamp is a type of amplifier using opamp where the output waveform will be phase opposite to the input waveform. The input waveform will be amplifier by the factor Av (voltage gain of the amplifier) in magnitude and its phase will be inverted.   13. What is an inverting amplifier used for?op-amp inverting amplifier. Op amp summing amplifier: Based around the inverting amplifier circuit with its virtual earth summing point, this circuit is ideal for summing audio inputs. It is widely used in audio mixer and many other applications where voltages need to be summed.   14. How does an inverting amplifier work?An inverting amplifier takes an input signal and turns it upside down at the op amp output. When the value of the input signal is positive, the output of the inverting amplifier is negative, and vice versa.   15. What is the gain of an inverting amplifier?The gain of inverting amplifier is Av= – Rf/Ri.

Introduction The audio power amplifier, also called power amp, is a sound-producing device that reconstructs the input audio signal on the output element. The reconstructed signal volume and power level must be idea, effective and low distortion. The audio frequency range is about 20Hz to 20000Hz, so the amplifier must have a good frequency response in this range. Depending on the application, the power level varies greatly, from the milliwatt level of earphones to the several watts of TV or PC audio, to the tens of watts of home stereo and car audio, to the more powerful home and commercial audio, even the system’s hundreds of watts are large enough to meet the sound requirements of the movie theater or auditorium. Catalog Introduction Ⅰ What is Audio Power Amplifier? Ⅱ Types of Audio Power Amplifiers 2.1 Class A Power Amplifier 2.2 Class B Power Amplifier 2.3 Class AB Power Amplifier 2.4 Class D Power Amplifier Ⅲ Audio Power Amplifiers Comparisons Ⅳ Audio Power Amplifier Circuit Architecture Ⅴ Typical Audio Amplifier Circuits Examples Ⅵ FAQ Ⅰ What is Audio Power Amplifier? Audio power amplifier is one of the important components of multimedia products and is widely used in the field of consumer electronics. Linear audio power amplifiers have always dominated the traditional audio amplifier market due to their low distortion and good sound quality. With the popularization of portable multimedia devices such as mobile phones, tablet PC, and notebook computers, the efficiency and volume of linear power amplifiers can meet the requirements of the market, for example, class D power amplifiers have become more and more popular for their advantages such as high efficiency and small size.The development of audio amplifier has experienced three generations of electron tube (vacuum tube), bipolar transistor and field effect tube. The tube audio amplifier has a mellow tone, but it is bulky, with the disadvantages of high power consumption, extremely unstable, and poor high-frequency response. Bipolar transistor audio amplifiers have bandwidth, large dynamic range, high reliability, long life, and high-frequency response, but its static power consumption and on-resistance are very large, so it is difficult to improve its efficiency. The FET audio amplifier has the same mellow tone as the electronic tube, and its dynamic range is wide, and more importantly, its on-resistance is small, which can achieve very high efficiency. audio power amplifier" width="410" height="250" /> Figure 1. Hi-Fi Audio Power Amplifier Ⅱ Types of Audio Power Amplifiers There are many types of audio power amplifiers, and more than ten modes of it commonly used, such as Class A, Class B, Class AB, Class C, Class D, Class E, Class F, Class G, Class H, and Class S, but only four types suitable for audio applications: Class A, Class B, Class AB and Class D. 2.1 Class A Power Amplifier The main feature of the Class A power amp is: The operating point Q of the amplifier is set near the midpoint of the load line, and the transistor is turned on during the entire cycle of the input signal. The amplifier can work in a single tube or push-pull state. Since the amplifier works in the linear range of the characteristic curve, transient and alternating distortion are small. The circuit is simple and the debugging is convenient. Class A power amplifiers only need one transistor to provide current to the load, and the conduction angle is 360°C during a complete signal cycle. This kind of power amplifier has low distortion, but often requires a large static current and is low in efficiency. Theoretically, the maximum working efficiency of a class A power amplifier is 25%, so a heat sink is required when working. 2.2 Class B Power Amplifier Class B power amplifier is composed of two complementary transistors. In a complete signal cycle, each amplifier tube will be turned on in one half cycle and turned off in the other half cycle, that is, the conduction angle is only 180°C. Because it has no static current, it has a higher efficiency. In theory, the maximum efficiency of a Class B power amplifier can reach 78%. However, when the input signal is close to zero, the amplifier tube has a critical conduction state, resulting in crossover distortion. 2.3 Class AB Power Amplifier Class AB power amplifier inserts two diodes into the input end of the Class B power amplifier. When the input is close to zero, the amplifier tube has been slightly turned on, so that the conduction angle of each amplifier tube is greater than 180°C and less than 360°C. It overcomes the crossover distortion of Class B power amplifiers, and the efficiency is between Class A and Class B power amplifiers, based on a common structure of traditional linear power amplifiers. However, the medium output voltage is usually far away from the power supply voltage, and a lot of power consumption is consumed in the transistor. Therefore, even a well-designed amplifier, its efficiency is still low. 2.4 Class D Power Amplifier Class D (digital audio power) power amplifier is a kind of input analog audio signal or PCM digital information into PWM (pulse brightness modulation) or PDM (pulse density modulation) pulse signals, which is used to control the amplifier power switching devices turn on/off audio power amplifiers, also known as switching amplifiers. It has the outstanding advantage of high efficiency.The output stage of the Class D power amplifier consists of two complementary power tubes. Driven by the high-frequency control pulse signal, the power tubes work in the on-off state. One is turned on and the other is turned off. Therefore, the amplifier does not require static power consumption, that is, it has a very high efficiency. In theory, the efficiency of Class D power amplifiers can reach 100%, but in fact it is over 80%, which is 2 to 3 times that of traditional linear power amplifiers.The digital audio power amplifier also looks like a one-bit power digital-to-analog converter. The amplifier is composed of four parts: input signal processing circuit, switching signal forming circuit, high-power switching circuit (half-bridge and full-bridge) and low-pass filter (LC). In electronics, Class D amplifiers use a very high frequency switch circuit to amplify the audio signal. DIY Class D Audio Amplifier Class D amplifiers have the following advantages:1) It has a high efficiency, usually above 85%.2) Small size, which can save a lot of space than analog amplifier circuits.3) Connection without crack noise.4) Have low distortion and good frequency response curve. Few peripheral components, easy to design and debug.   Ⅲ Audio Power Amplifiers Comparisons 1) Class A, B, and AB amplifiers are analog amplifiers, and class D amplifiers are digital amplifiers. 2) Class B and Class AB push-pull amplifiers have higher efficiency and less distortion than Class A amplifiers, and their transistors consume less power and have better heat dissipation. However, Class B amplifiers will have poor switching characteristics during the transition between transistor on and off states or cause alternate distortion due to improper selection of circuit parameters. 3) The Class D amplifier has high efficiency and low distortion, a good frequency response curve, and fewer peripheral components. 4) Class AB amplifier and Class D amplifier are the basic circuit forms of audio power amplifiers at present. Figure 2. Amplifier Circuit Example Ⅳ Audio Power Amplifier Circuit Architecture The purpose of audio amplifier is to reproduce the audio input signal with high efficiency and low distortion on the sound output element at the required volume and power level. The frequency range of the audio signal is 20 Hz to 20000 Hz, so the audio amplifier must have a good frequency response. Audio amplifiers usually consist of preamplifiers and power amplifiers.PreamplifierThe amplitude of the audio signal source signal is generally very small and cannot directly drive the power amplifier. Therefore, they must be amplified to a certain condition first, which requires the use of a preamplifier. In addition to the signal amplification function, it can also have functions such as volume adjustment, tone control, loudness control, and channel equalization.Power amplifierThe power amplifier is referred to as the power amplifier for short, and its purpose is to provide the load with a large enough current drive capability to achieve power amplification. Class D power amplifier works in the on-off state. In theory, it does not require static current and has high efficiency.The topological structure of a typical Class D power amplifier circuit is shown in Figure 2: It consists of a triangle wave generator, a comparator, a power output stage and an LC low-pass filter. Figure 3. Class D Power Amplifier Circuit Figure 2 is a schematic diagram of the working principle of a Class D power amplifier. The sine wave audio input signal and the triangular wave signal with a much higher frequency are modulated by the comparator to obtain a PWM modulation signal whose duty cycle is proportional to the amplitude of the input signal. It pushes the output power tube to work in the on-off state. The output end of the tube obtains an output signal with a constant duty cycle. The amplitude of the output signal is the power supply voltage and has a strong current drive capability. After signal modulation, the output signal contains both the input signal and the fundamental component of the modulated triangle wave, as well as their higher harmonics and combinations. After LC low-pass filtering, the high-frequency components in the output signal are filtered out, and a low-frequency signal with the same frequency and amplified amplitude as the original audio signal is obtained on the load.   Ⅴ Typical Audio Amplifier Circuits Examples The following is a brief introduction of AN7115 audio power amplifier circuit. Figure 4. AN7115 Audio Amplifier Circuit AN7115 has an output power of 2.1W and a noise output of 3mV under the conditions of V=9.0V, THD=10%, and RL=8Ω.Limit parameters: Vcc=13V, power dissipation (without radiator) is 1.2W, and 2.25W with a radiator. Working temperature: -20℃ to 70℃, suitable for small portable radio recorders and audio equipment as power amplifiers. TDA2030 adopts V-shaped 5-pin single in-line plastic package structure. According to the shape of the pin, it can be divided into H-type and V-type. This integrated circuit is widely used in car stereo radio and tape recorders and mid-power audio equipment. It has the characteristics of small size, high output power, low distortion, etc., and has an internal protection circuit. Circuit characteristics are as followed: Figure 5. TDA2030 Audio Power Amplifier Circuit 1) Very few external components.2) The output power is large, Po=18W (RL=4Ω).3) The use of ultra-small package (TO-220) can increase the assembly density.4) The boot impact is minimal.5) It contains various protection circuits, so it is safe and reliable to work. The main protection circuits include: short circuit protection, thermal protection, ground wire coupling open circuit, power supply polarity reverse connection (Vsmax=12V), and load discharge voltage kickback, etc. Figure 6. Mini Audio Power Amplifier Ⅵ FAQ 1. What is audio power amplifier?An audio power amplifier (or power amp) is an electronic amplifier that amplifies low-power electronic audio signals such as the signal from radio receiver or electric guitar pickup to a level that is high enough for driving loudspeakers or headphones. Audio power amplifiers are found in all manner of sound systems including sound reinforcement, public address and home audio systems and musical instrument amplifiers like guitar amplifiers.   2. How do audio amplifiers work?An amplifier takes an input signal from a source, such as a laptop, turntable or CD player, and creates a larger copy of the original signal before it's sent to the speakers. It gets the power to do this from your mains electricity, which is sent directly to the power supply within the amplifier.   3. What does an audio power amplifier do?An audio power amplifier (or power amp) is an electronic amplifier that amplifies low-power electronic audio signals such as the signal from radio receiver or electric guitar pickup to a level that is high enough for driving loudspeakers or headphones.   4. What is the most powerful audio amplifier?Classic audio brand McIntosh has announced its most powerful integrated amplifier. The fully loaded, hybrid MA12000 Integrated Amplifier is McIntosh's most powerful integrated amplifier ever offering 350W per channel.   5. Do you need amplifier for speakers?Powered speakers do not need an amplifier. They have an amplifier already installed in them which is why they are called 'powered speakers' to begin with. Depending on the speakers' input options, you can hook them up to different audio sources without the need for a separate amplifier.   6. Does an amplifier improve sound quality?Amplifiers ideally amplify audio signals linearly and, therefore, do not technically improve or worsen sound quality. However, less-than-ideal amplifiers, amp settings and amplifier-speaker combinations may worsen sound quality. Amps are nevertheless needed to drive speakers and headphones properly.   7. Why do I need a power amplifier?Some users prefer to separate input switching and AV processing from the task of providing power for, and connection of, loudspeakers through separate AV preamp/processors and power amplifiers. ... A separate preamp and power amp results in more equipment and cable clutter.   8. Which is better power amplifier or integrated amplifier?A preamplifier / power amp does generally offer better quality; while flat output has become common place, keeping your switching circuitry separate from your amplification and having two separate power supplies can reduce the internal interference that can be introduced into your signal.   9. What is a home audio amplifier?A home theater amplifier (also known as an audio-visual receiver or simply an AVR) is a powerful piece of kit that combines amplifiers and digital signal processors to convert output from a range of source devices into high quality sound and video. In short, it is the hub of a home theater set up.   10. What is the most powerful audio amplifier?Classic audio brand McIntosh has announced its most powerful integrated amplifier. The fully loaded, hybrid MA12000 Integrated Amplifier is McIntosh's most powerful integrated amplifier ever offering 350W per channel.   11. What is the purpose of an audio amplifier?The goal of audio amplifiers is to reproduce input audio signals at sound-producing output elements, with desired volume and power levels—faithfully, efficiently, and at low distortion.   12. What do you mean by audio amplifier?Any electronic device that increases the power of an electrical signal whose vibrations are confined to the audio frequency range—the range that can be perceived by the human ear—is an audio amplifier.

IntroductionWith the advancement of automobile technology, an increasing number of cars are equipped with car tire pressure monitoring systems, which provide us with a quick and real-time understanding of the car tire pressure, and we no longer have to worry about insufficient and excessive tire pressure before driving, and the car tire pressure monitoring system has brought great convenience to our car. Currently, most car tires are equipped with pressure sensors to detect pressure changes in order to ensure the safety of car driving. According to relevant statistics, tire pressure reaches a reasonable value, which can not only improve driving safety but also reduce fuel consumption. So, how does a car tire pressure sensor work? This article will introduce in detail.Video: Tire Pressure Sensors CatalogIntroductionⅠ Principles of Tire Pressure Sensors in AutomobilesⅡ Design Background of TPMSⅢ Tire Pressure Sensors in a Tire Pressure Monitoring SystemⅣ Details of Tire Pressure Sensors in Direct Tire Pressure Monitoring SystemⅤ SummaryⅥ Frequently Asked Questions about Tire Pressure Sensors Ⅰ Principles of Tire Pressure Sensors in Automobiles1.Strain Sensors in AutomobilesThe principle of a strain sensor is primarily based on the resistive strain effect,when the conductor undergoes mechanical deformation due to external action, a corresponding change in resistance value occurs. Calculate the required pressure by first calculating the magnitude of strain using the relationship between the change in resistance value and the change in output electrical signal. The strain gauge pressure sensor is primarily used to measure the dynamic or static pressure of a flowing medium, such as the inlet and outlet gas or liquid pressure of power pipeline equipment, engine pressure, internal combustion engine pipeline pressure, and so on.The most widely used strain gauge is the paste strain gauge (strain gauge). Its main disadvantages are small output signal, narrow linear range, and poor dynamic response (see resistance strain gauge, semiconductor strain gauge). However, due to the small size of strain gauge, many specifications of commercial strain gauge can be selected, and the form of elastic sensor can be flexibly designed to adapt to various applications, strain type pressure sensor made by strain gauge is still widely used. According to the different structure of elastic sensor, strain type pressure sensor can be roughly divided into strain tube type, diaphragm type, strain beam type and combined type.Figure:Strain Sensors in Automobiles 2.Piezoresistive Pressure Sensors in AutomobilesThe piezoresistive pressure sensor's pressure sensitive element is a piezoresistive element that operates on the piezoresistive effect. The term "piezoresistive element" actually refers to the diffusion resistance created by integrated circuit technology on a semiconductor substrate. When subjected to external force, its resistance changes due to resistivity change. During normal operation, diffusion resistors must be attached to elastic elements, and monocrystalline silicon diaphragms are commonly used.The piezoresistive pressure sensor's main advantages are its small size, relatively simple structure, good dynamic response, high sensitivity, and ability to measure micro pressures of more than ten Pascals. It is a relatively ideal one, and it is currently being developed and applied at a rapid pace. Sensor of pressure. Non-linearity and temperature affect the measurement accuracy of this sensor, affecting the size of the piezoresistive coefficient. Microprocessors are used in today's intelligent piezoresistive pressure sensor to compensate for nonlinearity and temperature. It integrates the sensor and computer on the same silicon chip using large-scale integrated circuit technology, and it has functions such as signal detection, processing, and memory. As a result, the sensor's stability and measurement accuracy are greatly improved.Figure:Piezoresistive Pressure Sensor 3.Other Automotive Pressure SensorsFurthermore, differential transformer type pressure sensors (LVDT) and surface elastic wave pressure sensors are available (SAW). SAW type pressure sensors have small size, light weight, low power consumption, high reliability, high sensitivity, high resolution, digital output, and so on. LVDT type pressure sensors have larger output, easy to digital output, but poor anti-interference. It is used to detect the pressure of an automobile suction valve and can operate reliably in high temperatures. Ⅱ Design Background of TPMSThe tire pressure influences both the car's excellent performance and the length of the tire's service life. According to SAE (Society of Automotive Engineers) data, there are more than 260,000 traffic accidents caused by tire failure in the United States each year, with flat tires accounting for 70% of highway accidents. Furthermore, natural tire leakage or under-inflation is the leading cause of tire failure. Every year, approximately 75% of tire failures occur. The data also shows that in high-speed driving, punctures caused by tire failure are a major cause of traffic accidents.Flat tires, the unseen killer, have caused numerous human tragedies and incalculable economic losses to the country and businesses. As a result, in order to reduce the number of traffic accidents caused by flat tires, the US federal government has mandated that automakers accelerate the development of TPMS ( Tire Pressure Monitoring System).Figure:Tire Pressure Monitoring System Ⅲ Tire Pressure Sensors in a Tire Pressure Monitoring SystemThere are two main tire pressure monitoring system solutions: direct system and indirect system.The direct tire pressure monitoring system measures tire pressure directly using the pressure sensor installed in each tire and displays and monitors the tire pressure. The system will automatically alert if the tire pressure is too low or if there is a leak.To monitor tire pressure, the indirect tire pressure monitoring system compares the speed difference between tires using the wheel speed sensor of the automobile abs system.The main disadvantages of this type of system are as follows: 1. The accurate instantaneous air pressure value of each tire cannot be displayed; 2. It is not possible to alarm when the pressure of the same axle, same side wheel, or all tires drops at the same time; and 3. Factors such as vehicle speed and detection accuracy cannot be taken into account simultaneously.Figure:Tire Pressure Sensors in a TPMS Ⅳ Details of Tire Pressure Sensors in Direct Tire Pressure Monitoring SystemThere are two types of direct tire pressure monitoring systems: active and passive.The active system makes capacitive or piezoresistive pressure sensors on silicon using mems technology. Each rim has a pressure sensor, and the signal is transmitted via radio frequency. It is set up in the wireless cab. The pressure sensitive signal is received by the receiving device, which then displays the current tire pressure after some signal processing.The advantage of active technology is that it is relatively mature, and the modules developed can be applied to tires of various brands, but the disadvantages are also more noticeable. Its induction module requires battery power, posing a problem with system service life.The passive tire pressure monitoring system's sensor is based on surface acoustic waves. A radio frequency electric field is used to generate a surface acoustic wave in this sensor. The surface acoustic wave changes as it passes through the surface of the piezoelectric substrate material. The change in the surface acoustic wave can indicate tire pressure. Although this technology does not require battery power, it does necessitate the integration of the transponder into the tire, and it can only be implemented if tire manufacturers agree on a common standard. The tire pressure monitoring system must detect abnormal tire pressure conditions, and it can only do so with high resolution and accuracy. Battery life is limited, and capacity is affected by temperature as well. It is best for the sensor to perform passive detection in order to improve system reliability. According to studies, the information collected by tire pressure sensors can be used to monitor vehicle suspension failures and correct navigation systems. As a result, the future automotive pressure sensor should be a passive intelligent sensor with multiple functions. Ⅴ SummaryIn recent years, automotive sensors have been the fastest growing and most widely used sensor category. The growth of the automotive industry encourages the rapid development of automotive pressure sensors. Sensor performance is improving as a result of advances in manufacturing and process technology, and tire pressure monitoring is becoming increasingly important. Ⅵ Frequently Asked Questions about Tire Pressure Sensors1.When should a tire pressure sensor be replaced?TPMS sensors are designed to last for many years.5-10 years is a likely lifespan. Given their cost, most drivers will be inclined to replace TPMS sensors on an “as needed” basis.In other words, only once their batteries have expired, or other TPMS components have failed. 2.Is it safe to drive with tire pressure sensor fault?No, driving with the TPMS Light on is not safe. It means one of your tires is underinflated or overinflated. This can cause undue wear on the tire, potentially lead to a tire failure, and cause a blowout dangerous to you and other drivers on the road. 3.How much does it cost to replace a tire pressure sensor?The average cost for TPMS sensor replacement is between $207 and $257. Labor costs are estimated between $53 and $67 while parts are priced between $154 and $190. This range does not include taxes and fees, and does not factor in your specific vehicle or unique location. Related repairs may also be needed. 4.How do you fix a tire pressure sensor?Without starting the car, turn the key to the “On” position. Press the TPMS reset button and hold it until the light blinks three times, then release it. Start the car and let it run for 20 minutes to reset the sensor. You'll usually find the tire pressure monitor reset button beneath the steering wheel. 5.How do I know if my tire pressure sensor is bad?A small light illuminates at the dashboard's display panel whenever there is a problem with the tire pressure sensors. It appears as a vivid yellow exclamation point inside of a U symbol; you'd be easily able to spot it. As soon as it turns on, the driver must check the tire for less or no air.  

IntroductionPrinted Circuit Board(PCB) is a board of most modern electronic devices that has lines and pads that connect various points together. Even if it is a small board, its manufacturing process is very cumbersome and exquisite. Here will introduce the PCB manufacturing process steps by steps with pictures and video.How is PCB made?The following are the detailed PCB producing processes:IntroductionStep 1. PCB CAD FileStep 2. Plate ProductionStep 3. PCB Inner LayersStep 4. Board Punching and CheckingStep 5. LaminationStep 6. DrillStep 7. Copper Chemical Precipitation on the HolesStep 8. PCB Outer LayersStep 9. Computer Control and Copper ElectroplatingStep 1. PCB CAD FileThe first step in PCB production is to organize and check the PCB layout. The PCB manufacturers get the CAD files from the PCB design company, and they will convert them into a unified format-Extended Gerber RS-274X or Gerber X2, because each CAD software has its own unique file format. Then the electronic engineers will check whether the PCB layout conforms to the manufacturing process, and whether there are any defects and other issues.Figure 1. PCB CAD FileWhen making a PCB at home, the PCB layout can be printed on paper with a laser printer, and then transferred to the copper clad laminate. During the printing process, because the printer is prone to lack of ink and breakpoints, it is necessary to manually fill up the ink with an oil-based pen. Figure 2. PCB Laser PrintingHowever, the factory generally uses photocopying to print the PCB layout on the film. If it is a multi-layer PCB, the layout film photocopied on each layer will be arranged in order.Figure 3. PCB Film Arranged in OrderThen the film will be punched with alignment holes. Alignment holes are very important, which is essential to align the materials of each layer of the PCB.Step 2. Plate ProductionClean the copper plate. If there is dust, it may cause the final circuit to be short-circuited or broken.Figure 4. Clean the Copper PlateThe figure below is an example of an 8-layer PCB, which is actually made up of 3 copper clad laminates plus 2 copper films, and then glued them together with prepregs. The production sequence is to start with the middle board (4th- and 5th-layer of circuits), continuously stack together, and then fix. The production of 4-layer PCB is similar, including one core board and two copper films.Figure 5. 8-layer PCB Plate DisplayStep 3. PCB Inner LayersFirst, make the two-layer circuit of the middle core board. After the copper clad laminate is cleaned, it will be covered with a photosensitive film on the surface. This film will solidify when exposed to light, forming a protective film on the copper foil.Figure 6. PCB CoreInsert the two-layer PCB layout film and the double-layer copper clad laminate into the upper PCB layout film to ensure that the upper and lower PCB layout films are stacked accurately.Figure 7. PCB Layout Film PlacingThe machine irradiates the photosensitive film on the copper foil with a UV lamp. The transparent film is cured under the light, and there is still no cured photosensitive film. The copper foil covered under the cured film is the required PCB layout, which is equivalent to the function of the laser printer ink of the manual PCB. In addition, the copper foil covered by the black film will be corroded away, and the cured transparent film will be preserved.Figure 8. Cured Photosensitive FilmClean the uncured photosensitive film with lye, and the required copper foil circuit will be covered by the cured film.Figure 9. Clean Uncured Photosensitive FilmThen use a strong base, such as NaOH, to etch away the unnecessary copper foil.Figure 10. Copper Foil EtchingTear off the cured photosensitive film to expose the copper foil of the required PCB layout.Figure 11. Tear Off the Cured Photosensitive FilmStep 4. Board Punching and CheckingThe core board has been successfully produced. Then punch alignment holes on it to facilitate with other materials.Figure 12. Punch Alignment Holes on PCBOnce the core board is pressed together with other layers, it cannot be modified. So PCB checking is very important. The machine will automatically compare with the PCB layout drawing to find out the error.Figure 13. PCB Layout Drawing ComparisonThe first two layers of PCB boards have been made.Step 5. LaminationA new raw material is introduced here called Prepreg, which is the adhesive among the core boards(PCB layers>4), as well as the core board and the outer copper foil, and it also plays a role in insulation.Figure 14. PCB Prepreg and CopperThe lower copper foil and the two layers of prepreg have been fixed in advance through the alignment hole and the lower iron plate, and then the finished core board is also placed in the alignment hole, and finally the two layers of prepreg, a layer of copper foil and a layer of pressure-bearing aluminum plate covers the core plate.Figure 15. Fixed PCB Prepreg and CopperIn order to improve work efficiency, this factory will stack three different PCB boards together before fixing them. The upper iron plate is magnetically attracted to facilitate alignment with the lower iron plate. After the two layers of iron plates are successfully aligned by inserting the alignment pins, the machine compresses the space between the iron plates as much as possible, and then fixes them with nails.Figure 16. Fixed PCB LayersThe PCB boards clamped by the iron plates are placed on the holder, and then sent to the vacuum heat press for laminating. The high temperature can melt the epoxy resin in the prepreg and fix the core boards and copper foils together under pressure.Figure 17. PCB Layers LaminationAfter the lamination, remove the upper iron plate that presses the PCB. Then remove the pressure-bearing aluminum plate. The aluminum plate also plays the role of isolating different PCBs and ensuring the smoothness of the outer copper foil of the PCB. Finally the PCB taken out at this time will be covered by a layer of smooth copper foil.Figure 18. Remove the Upper Iron Plate and Aluminum PlateStep 6. DrillSo how to connect 4 layers of copper foils that are not in contact with each other in the PCB? First, make the through-hole through the PCB, and then metalize the hole wall to conduct electricity.Figure 19. PCB DrillPut a layer of aluminum plate on the punching machine, and then put the PCB on it. Since drilling is a relatively slow process, in order to improve efficiency, according to the number of layers of the PCB, 1 to 3 identical boards are stacked for drilling together. Finally, cover the uppermost PCB with a layer of aluminum plate. The upper and lower of aluminum plates are used to prevent the copper foil on the PCB from tearing when drilling.Figure 20. PCB DrillNext, you only need to select the correct drilling program on the computer, and the rest is done automatically by the drilling machine. The drill bit is driven by air pressure, and the maximum rotation speed can reach 150,000 revolutions per minute. Because such a high rotation speed is sufficient to ensure the smoothness of the hole wall.Figure 21. Drill ProgramThe replacement of the drill bit is also automatically completed by the machine according to the program. The smallest drill bit can reach a diameter of 100 microns, while the diameter of a human hair is 150 microns.Figure 22. Drill ReplaceIn the previous process, the molten epoxy was squeezed out of the PCB, so it needed to be cut off. Here the profiling milling machine cuts its periphery according to the correct XY coordinates of the PCB.Figure 23. Cuts PCB PeripheryStep 7. Copper Chemical Precipitation on the HolesSince almost all PCB designs use perforations to connect different layers of lines, a good connection requires a 25-micron copper film on the hole wall. The thickness of the copper film needs to be realized by electroplating, but the hole wall is composed of non-conductive epoxy resin and glass fiber board. So the first step is to deposit a layer of conductive material on the hole wall, and form a 1 micron copper film on the entire PCB surface by chemical deposition. The entire process such as chemical treatment and cleaning is controlled by the machine.Step 8. PCB Outer LayersNext, the PCB outer layer is transferred to the copper foil. The process is similar to the transfer principle of the previous PCB inner core board. The PCB layout is transferred to the copper foil by photocopying film and photosensitive film. The only difference is positive films will be used as boards.The transfer of the internal PCB layout described above uses the subtractive method, and the negative film is used as the board. The PCB is covered by the cured photosensitive film as a circuit, and the uncured film is cleaned. After the exposed copper foil is etched, the PCB layout circuit is protected by the cured film. The transfer of the outer PCB layout adopts the normal method, and the positive film is used as the board. The non-circuit area is covered by the cured photosensitive film on the PCB. After cleaning the uncured film, electroplating is performed. Where there is a film, it cannot be electroplated, and where there is no film, copper is plated first and then tin is plated. After the film is removed, alkaline etching is performed, and finally the tin is removed. So the circuit pattern remains on the board because it is protected by tin.Put the cleaned PCB on both sides of the copper foil into the laminating machine, and the photosensitive mold will be pressed onto the copper foil.Figure 24. LaminatorFix the printed PCB layout film of the upper and lower layers through the holes, and put the PCB board in the middle. Then, the photosensitive film under the light-transmitting film is cured by the irradiation of the UV lamp, which is the circuit that needs to be reserved.Figure 25. PCB Expose to the UV LightAfter cleaning off the unnecessary and uncured photosensitive film, inspect the PCB board.Figure 26. PCB CheckingClamp the PCB with clips, and electroplate the copper. As mentioned earlier, in order to ensure that the holes have sufficient conductivity, the copper film plated on the hole walls must have a thickness of 25 microns, so the entire system will be automatically controlled by the computer to ensure its accuracy.Figure 27. PCB Copper PlatingStep 9. Computer Control and Copper ElectroplatingAfter the copper film is electroplated, the computer gives instructions to electroplate a thin layer of tin. Then, check to ensure that the thickness of the plated copper and tin is correct.Figure 28. Electroplated Copper and Tin InspectionNext, a complete automated assembly line completes the etching process. Then, clean the cured photosensitive film on the PCB.Figure 29. Clean Cured Photosensitive FilmThen use a strong alkali to clean the unnecessary copper foil covered by it.Figure 30. Clean the Unnecessary Copper FoilFinally, use the tin stripping solution to strip the tin plating on the PCB layout copper foil. After cleaning, the 4-layer PCB layout is complete. Frequently Asked Questions about PCB Manufacturing Process1. Which are the techniques of PCB manufacturing?There are several PCB manufacturing methods that a PCB can be submitted to before reaching the final product. These methods include preparing the board's surface, placing components, soldering, cleaning, and inspection and testing. 2. What is PCB design process?Step 1 – The DesignStep 2 – Printing the DesignStep 3 – Creating the SubstrateStep 4 – Printing the Inner LayersStep 5 – Ultraviolet LightStep 6 – Removing Unwanted CopperStep 7 – Inspection.Step 8 – Laminating the Layers 3. Which software is best for PCB design?Top 8 Best PCB Design Software of 2021PROTEL (Altium Designer)PADS (PowerPCB)ORCADAllegroEagle (Easily Applicable Graphical Layout Editor)KicadEasyEdaFritzing 4. What is a PCB layer?A PCB is defined as a number of copper layers in a well defined sequence. Copper layers of a PCB are usually just named layers or also called SIGNAL layer. However, to define the complete PCB, other layers are required. They are usually named by their functionality and position. 5. What are the components of a PCB?Some common PCB components include:Battery: provides the voltage to the circuit.Resistors: control the electric current as it passes through them. They’re colour coded to determine their value.LEDs: light emitting diode. Lights up when current flows through it, and will only allow current to flow in one direction.Transistor: amplifies charge.Capacitators: these are components which can harbour electrical charge.Inductor: stores charge and stops and change in current.Diode: allows current to pass in one direction only, blocking the other.Switches: can either allow current or block depending if they are closed or open.

IntroductionAs everyone knows, in order to create a passive low pass filter, combing resistive elements with reactive elements happens often. Put simply, a typical circuit composed of resistors and capacitors or inductors. According to theories, the resistor–inductor (RL) low-pass topology is equivalent to the resistor-capacitor (RC) low-pass topology in terms of filtering capability. However. in fact, RC low pass filters are more common, so this article will focus on first-order RC low pass filters.In this video, Passive RC Low Pass Filter has been discussed. CatalogIntroductionⅠ Typical RC Circuit1.1 Time Domain1.2 Frequency DomainⅡ First-order Low Pass Filter on Software2.1 Basic Filtering Algorithm2.2 Basic Algorithm of First-order RC Digital FilteringⅢ Optimization Method- Filtering Coefficients AdjustmentⅠ Typical RC CircuitThe RC circuit has thousands of uses and is a very important circuit to study. Not only can it be used to time circuits, it can also be used to filter out unwanted frequencies in a circuit and used in power supplies, like the one for your computer, to help turn ac voltage to dc voltage.Figure 1. Typical RC Circuit (DC, AC, and Pulse Signals can all use it)1.1 Time DomainCapacitor Current:According to Kirchhoff’s Voltage Law:Where, the unit of Ui is volts, the unit of RC is seconds, and τ=RC, get:Suppose the initial voltage of the capacitor is 0, where:R=1000ΩC=4.7uFUi=1Vt=0.0001~0.1sτ=RCVc(τ)=0.632 Figure 2. Step Response Curve of a First-order RC System1.2 Frequency DomainTaking the capacitor voltage as the output, the network function of the circuit is:Where u1=Ui, u2=UoLet ωc be equal to:, which is the cut-off frequency.Amplitude and phase angle function:Value of variables:R=1000ΩC=4.7uF |A(fc)|=0.707θ(fc)=-45, f=0.001, 1, …….100000.Amplitude and phase frequency characteristics:Figure 3.Figure 4.Logarithmic representation of amplitude-frequency characteristic:Figure 5.Analysis:When ω<ωc, the amplitude is a straight line parallel to the coordinate, and there is no attenuation.　When ω>ωc, it is a straight line whose slope is proportional to -20dB/decade.When ω=ωc, the gain is attenuated to 0.707, which is -3dB, and the phase lags by 45 degrees, corresponding to a low-pass filter. This frequency is usually called the cutoff frequency. Disadvantages:When using this analog filter to suppress low-frequency interference, the filter is required to have a larger time constant and a high-precision RC network. Increasing the time constant requires increasing the value of R, and meanwhile, the leakage current increases accordingly, thereby reducing the filtering effect.Figure 6. RC CircuitⅡ First-order Low Pass Filter on SoftwareAdvantages1) The use of digital filtering algorithms to achieve dynamic RC filtering can well overcome the shortcomings of analog filters.2) This kind of algorithm is more practical when the simulation constant is required.3) It has a good inhibitory effect on periodic interference.4) Save RAM space Disadvantages1) Exit phase lag, resulting in low sensitivity.2) It cannot filter out interference with a frequency higher than half of the sampling frequency (called the Nyquist frequency. For example, if the sampling frequency is 100 Hz, it cannot filter out interference signals above 50Hz). In this case, an analog filter should be used.3) For the single-chip microcomputer without multiplication and division running instructions, the workload of the program operation is relatively large.2.1 Basic Filtering AlgorithmOrigin of the AlgorithmThe transfer function of the first-order RC low-pass filter in the S domain for frequency analysis:Through z-transformation (there are many methods, such as first-order forward difference, bilinear transformation, etc. Here, the first-order backward difference method is used): Into the S-domain Transfer Function After the derivation is transformed into the difference equation, we can get:The transfer function in the S domain can be transformed into a difference equation in the time domain through the Z transformation.2.2 Basic Algorithm of First-order RC Digital FilteringX is the input, Y is the output value after filtering, then: a is a parameter related to the RC value, called the filter coefficient, its value determines the weight of the new sample value in the filtering result of this time, and its value is usually far less than 1, when the sampling interval t is small enough:1) The smaller the filtering coefficient, the smoother the filtering result, but the lower the sensitivity.2) The larger the filtering coefficient, the higher the sensitivity, but the more unstable the filtering result.3) The output value this time mainly depends on the last filtered output value, and the current sampled value has a relatively small effect on this output, which plays a corrective role.4) Cutoff frequencyFor example: t=0.5s (f=2Hz), a=1/32where fl=(1/32)/(2*3.14*0.5)=0.01Hz Basic ProgramWrite the program according to the basic principles and formulas of first-order filter, as follows:/*In the program, integer arithmetic is faster than decimal arithmetic. In order to speed up the processing speed of the program, for calculation convenience, a is an integer (from 0~255), 1-a is replaced by 256-a, which means that the new sample value is being filtered. The weight in the result (you can also change the base of 1-a to 100-a, and the calculation result will be processed accordingly)*/#define a 128 char value; //Last filtering valuechar filter(){    char new_value;    new_value=get_ad();//Sampling value    return(256-a)*value/256+a*new_value/256；}Initial Optimization of the ProgramReduce the number of operations of multiplication and division to increase the speed of operations.Specific optimization methods:First compare the new sampled value with the previous filtering result, and then use different formula calculations based on the comparison, so that the calculation efficiency of the program is doubled.Resolve the basic formula to get: ProcessNotes:S → New Sampling ValueR → Previous Filtering ResultC→ Filter CoefficientN→ New Filtering Result Program/*Int: NEW_DATA     New sampling values       OLD_DATA       Last filtering result       k        Filter coefficient (0~255)  Out:         The filtering results */ char filter_1(char NEW_DATA,char OLD_DATA,char k){    int result;    if(NEW_DATA<OLD_DATA)    {        result=OLD_DATA-NEW_DATA;        result=result*k;        result=result+128;//+128 Round Up        result=result/256;        result=OLD_DATA-result;    }    else if(NEW_DATA>OLD_DATA)    {        result=NEW_DATA-OLD_DATA;        result=result*k;        result=result+128;//+128 Round Up        result=result/256;        result=OLD_DATA-result;    }    else result=OLD_DATA;    return((char)result);} Filtering AnalysisWhen the filtering coefficient is 30:Figure 7.When the filtering coefficient is 128:Figure 8.When the filtering coefficient is 200:Figure 9.It can be seen that the smaller the filtering coefficient, the smoother the filtering result, but the lower the sensitivity. On the contrary, the larger the filtering coefficient, the higher the sensitivity, but the more unstable the filtering result.Insufficient1) The contradiction between sensitivity and smoothness2) Errors caused by discarding decimals.For example: the current sampling value=25, the last filtering result=24, and the filtering coefficient=10;According to the algorithm, the filtering result of this time = 24.0390625In single-chip microcomputers, floating-point numbers are rarely used, and the fractional part is either discarded or needs to round up. In this way, the result is 24. If the sampling value is always 25, the result will always be 24. Because the filtering result and the actual data will always have an error that cannot be eliminated. Sometimes it will cause the filtering result curve to deviate from the actual value when the sampling data is stable at a certain value (that is, there is a large error between the filtering result and the actual result although in a stable case). Be Careful1) Changing the filtering coefficient, increasing it will reduce the smoothness, and if it is too large, the filtering will lose its meaning.2) The use of decimal part in calculations will bring heavy computational pressure to the CPU. Ⅲ Optimization Method- Filtering Coefficients AdjustmentRealize the Function1) When the data changes rapidly, the filtering results can be followed up in time, and the faster the data changes, the higher the sensitivity should be (sensitivity priority principle).2) When the data becomes stable and oscillates within a range, the filtering result can become stable (the principle of stability first).3) When the data is stable, the filtering result can be approximated and finally equal to the sampling data (eliminate the error caused by decimals in the calculation). Judgment before Adjustment1) Whether the data changes consistently. For example, when the two consecutive sampling values are larger than the previous filtering result, it is normal, otherwise it is regarded as inconsistent.2) Whether the data changes quickly, which is to judge the difference between the sampling value and the previous filtering result.Adjustment Principle1) When the two data changes are inconsistent, it means there is jitter. Clear the filtering coefficient to zero, and delete the new sampling value.2) When the data changes consistently, gradually increase the filtering coefficient to provide the weight of this sampling.3) When the data changes quickly (difference value> debounce count acceleration response threshold), the filtering coefficient should be increased quickly. Adjusting Filter Coefficient Process① Calculate the difference (absolute value) between the current sampling value and the last filtering result; Set the data change direction flag.② Two changes in the same direction?③ First order filter coefficient + coefficient increment (the maximum value is taken when the result is greater than the maximum value). Several Constant Parameters and Their Ranges1. Debounce counting acceleration response threshold is determined according to the actual situation.2. The maximum value of debounce count, which is generally 10.3. The increment of filtering coefficient range is 10~30.4. The maximum value of the filtering coefficient is generally 255.Before starting the first-order filtering program, open the adjustment filter coefficient program to adjust the coefficients in real time. Filtering Effect1. When the sampled data is accidentally interfered, the interference in the filtering result is completely filtered out.2. When the data oscillates within a range, the filtering result curve is very smooth, almost a straight line.3. When the sampling data has real changes, the filtering results can be followed up in a relatively timely manner.4. When the sampling data becomes stable, the filtering result gradually approaches and is finally equal to it.Finally, improve the algorithm. Taking into account the requirements of sensitivity and stability; and meanwhile, it does not consume too much RAM space. As long as a few constants are adjusted reasonably, the algorithm is more suitable for practical applications. Frequently Asked Questions about RC Low Pass Filter1. What is RC low pass filter?A low pass filter is a filter which passes low-frequency signals and blocks, or impedes, high-frequency signals. ... Low pass filters can be constructed using resistors with either capacitors or inductors. A low pass filter composed of a resistor and a capacitor is called a low pass RC filter. 2. Why RC circuit is low pass filter?Then by carefully selecting the correct resistor-capacitor combination, we can create a RC circuit that allows a range of frequencies below a certain value to pass through the circuit unaffected while any frequencies applied to the circuit above this cut-off point to be attenuated, creating what is commonly called a rc low pass fiter. 3. What is difference between RC low pass filter and RC high pass filter?Low pass filter is the type of frequency domain filter that is used for smoothing the image. It attenuates the high frequency components and preserves the low frequency components. High pass filter: ... It attenuates the low frequency components and preserves the high frequency components. 4. What is the transfer function of a low pass filter?Low Pass Filters and their Transfer FunctionsAs its name implies, a low pass filter is an electronic device that allows low frequency AC signals to pass a current through the filter circuit. The output from the filter circuit will be attenuated, depending on the frequency of the input signal. 5. How is low pass filter frequency calculated?The cut-off frequency or -3dB point, can be found using the standard formula, ƒc = 1/(2πRC). The phase angle of the output signal at ƒc and is -45o for a Low Pass Filter.

IntroductionAn analog-to-digital converter, or A/D converter, or ADC for short, usually refers to an electronic device that converts an analog signal into a digital signal. Except for the most specialized analog-to-digital converters, all ADCs are implemented as integrated circuits (ICs). These are usually mixed-signal integrated circuit chips based on metal oxide semiconductor (MOS) that integrate analog and digital circuits.As we all know, ADC is mainly used to the digital acquisition of analog signals for for data processing purposes. The signals around us are generally continuously changing analog quantities, such as light, temperature, speed, pressure, sound, etc. However, most of us use digital equipment. If we want to use and process information easily, it is necessary to convert the analog quantity into a digital quantity and transmit it to the microcontroller or microprocessor. So how is ADC conversion realized? What kind of process is it? Reading the following note, you will definitely have a more comprehensive and systematic understanding of the analog-to-digital converter.What is ADC (Analog to Digital Converter)?CatalogIntroductionⅠ A/D Converter Basic1.1 Analog-to-Digital Converter Definition1.2 Analog to Digital Conversion Steps1.3 Why do We Need Analog-to-Digital Converter?Ⅱ Which A/D Converter is Better?Ⅲ What A/D Converter Includes?Ⅳ A/D Converter Applications and ICs4.1 Analog-to-Digital Converter Applications4.2 Analog-to-Digital Converter IC Modes ExplainedⅠ A/D Converter Basic1.1 Analog-to-Digital Converter DefinitionThe ADC converter is a system that converts analog signals into digital signals. It is a process of filtering, sample-and-hold, quantization and encoding. The analog signal passes band-limited filtering, sample-and-hold circuit, and becomes a ladder-shaped signal, and then passes through the encoder to make each level in the ladder-shaped signal become a binary code. Finally, the analog quantity is converted into a digital quantity and then transmitted to the CPU. That is to say, almost all energized data need ADC conversion. For example, electric energy metering of electric energy meters, weight measurement of electronic scales, temperature measurement of electronic thermometers, and communication fields.1.2 Analog to Digital Conversion StepsThe process of converting analog quantities into digital quantities is called analog-to-digital conversion, abbreviated as A/D, and the circuit that completes this function is called analog-to-digital converter, or ADC for short.Analog-to-Digital Conversion Steps Animation1) Sampling refers to replacing the original continuous signal in time with a sequence of signal samples at regular intervals, that is, discretizing the analog signal in time.2) Quantization uses a limited number of amplitude values to approximate the original continuously changing amplitude value, that is, changing the continuous amplitude of the analog signal into a limited number of discrete values with a certain interval.3) Encoding is based on a certain rule, the quantized value is represented by binary numbers, and then converted into a binary or multi-value digital signal stream. The digital signal obtained in this way can be transmitted through digital lines such as cables, microwave trunk lines, and satellite channels.The higher the signal frequency, the higher the operating frequency of the A/D circuit. The more digits, the more accurate the restoration accuracy of the signal. The I/O port of the MCU needs program cooperation to complete the A/D conversion. What’s more, the A/D chip can also be used alone to complete the analog-to-digital conversion.1.3 Why do We Need Analog-to-Digital Converter?Computer software, radio, and digital image acquisition all need the assistance of ADC converters, that is, the wave of human digitization has promoted the invention, development and continuous change of ADC converters. In short, the ADC converter plays an important role in human digitization.1) Many recording studios use 24-bit/96 kHz (or higher) pulse code modulation (PCM) or direct stream digital (DSD) recording formats, and then use ADC samples or decimates the signal for digital audio production on discs.2) Use ADC to store or transmit almost any analog signal in digital form. For example, TV tuner cards use fast video analog-to-digital converters. Digital storage oscilloscopes require very fast analog-to-digital converters, and ADCs are also crucial for software-defined radio and its new applications.3) Digital imaging systems usually use analog-to-digital converters to digitize pixels. Some radar systems usually use ADCs to convert signal strength into digital values for subsequent signal processing.4) Certain non-electronic or only partially electronic devices (such as rotary encoders) can also be regarded as analog-to-digital converters.Figure 1. Analog to Digital Conversion Example(Light Signal to Digital Signal) Ⅱ Which A/D Converter is Better?After years of development and continuous technological innovation, ADC converters have developed from Flash ADCs, Successive-Approximation ADCs, Counting/Slope Integration ADCs to sigma-delta (Σ-Δ) ADCs and Pipelined ADCs. They have their own advantages and disadvantages, and they can also meet different requirements.Successive-Approximation ADCs, Counting/Slope Integration ADCs and compression ADCs, etc. are mainly used in low-speed or medium-speed, medium-precision data acquisition and intelligent instruments. Hierarchical and pipelined ADCs are mainly used in high-speed signal processing, fast waveform storage and data recording, etc., such as video signal quantization and high-speed digital communication technology. ∑-△ ADC is mainly used in high-precision data acquisition, especially in electronic measurement fields such as digital sound systems, multimedia, seismic exploration instruments, sonar and so on. Here a brief description of the main ADC types is given below. Successive-Approximation ADCThe successive-approximation ADC is widely used. It includes a comparator, a digital-to-analog converter, a successive-approximation register (SAR) and a control logic unit. It is to continuously compare the sampling input signal with the known voltage. One clock cycle completes the 1-bit conversion, and the N-bit conversion requires N clock cycles. The conversion is completed and the output binary number is output. The resolution and sampling rate of this type ADC are contradictory: when the ADC resolution is low, the sampling rate is high, and if the resolution is to be improved, the sampling rate will be limited.Advantages: when the resolution is lower than 12 bits, the price is cheap, and the sampling rate can reach 1MSPS. Compared with other types, the power consumption is quite low.Disadvantages: In the case of higher than 14-bit resolution, the price is higher. The signal generated by the sensor needs to be conditioned before analog-to-digital conversion, including gain stage and filtering, so that the cost will increase significantly. Counting/Slope Integration ADCsCounting/Slope Integration ADC is also called dual-slope or multi-slope ADC, and its applications are also very wide. It is composed of an analog integrator with an input switch, a comparator and a counting unit. The input analog voltage is converted into a time interval proportional to its average value through two integrations. At the same time, a counter is used to count the clock pulses in this time interval, so as to realize the analog-to-digital conversion. Because the input end applies the integrator, it has a strong ability to suppress the interference of AC noise. For example, for high-frequency noise and fixed low-frequency (50Hz or 60Hz) interference suppression, it is suitable for use in noisy industrial environments. This type ADC is mainly used in low-speed, precision measurement and other fields, such as digital voltmeters.Advantages: High resolution, up to 22 bits; low power consumption and low cost.Disadvantages: The conversion rate is low, 100~300SPS at 12 bits. Parallel ADCsThe main feature of inter ADC is fast speed, which is the fastest of all types. The sampling rate can reach above 1GSPS. However, due to the limitations of power and volume, it is difficult to improve the resolution. The conversion of all bits of the ADC with this structure is completed at the same time, and the conversion time mainly depends on the switching speed of the comparator and the transmission time delay of the encoder. In addition, increasing the output code has little effect on the conversion time, but as the resolution increases, a high-density analog design  requires large number of precision divider resistors and comparator circuits for the conversion. That is to say, the output number is increased by one bit and the number of precision resistors is increased. It is about to double, and the comparator is also approximately doubled.The resolution of the parallel comparison ADC is limited by die size, input capacitance, power, etc. If the accuracy of the parallel comparators does not match, it will also cause static errors and increase the input offset voltage. Sigma-delta (Σ-Δ) ADCsThe Sigma-delta (Σ-Δ) ADC is composed of an integrator, a comparator, a 1-bit DA converter, and a digital filter. In principle, it is similar to the integral type. The input voltage is converted into a time (pulse width) signal and processed by a digital filter to obtain a digital value.Figure 2. Analog to Digital Converter Application ExampleⅢ What A/D Converter Includes?1) Sampling RateThe sampling rate indicates the rate at which the analog signal is converted into a digital signal, which is related to the manufacturing technology of the ADC device and depends on the judgment ability provided by the comparator in the ADC.Generally speaking, the sampling rate and resolution are mutually restrictive. Each time the sampling rate is doubled, the resolution losses 1bit. This is mainly due to the jitter during sampling, that is, aperture jitter or aperture uncertainty. 2) ADC ResolutionThe resolution indicates the number of bits after the analog signal is converted into a digital signal. It directly determines the quantization level of the ADC, that is, the minimum analog signal level value that the ADC can distinguish. Assuming that the ADC's input voltage range is (−V, V) and the resolution is N (bit), then the ADC has a 2N quantization level, so that the quantization level is: ΔV=2V/2N, where ΔV is the conversion accuracy. It can be seen from the above formula that the higher the resolution of the ADC and the smaller the voltage input range, the higher its conversion accuracy. 3) Signal-to-Noise Ratio (SNR)The signal-to-noise ratio (SNR) of the ADC reflects the ratio of the root mean square value of the noise-free signal part generated during the quantization process to the root mean square value of the quantization noise. If the input signal is a normalized sine wave 1/2sin(ωt+ψ), the SNR can be determined by the following formula: Among them, N is the resolution of ADC. It can be seen that the signal-to-noise ratio of the ADC mainly depends on the resolution. Every time the resolution increases by one bit, the SNR will increase by 6dB. However, as the resolution increases, the quantization level of the ADC becomes smaller, and the sampling process is more likely to be disturbed. 4) Effective Number of Bits (ENOB)ENOB is a measure of the dynamic range of an ADC converter. For the actual A/D conversion system, due to the influence of factors such as electrical noise, external interference, and non-linear distortion of analog circuits, it is not enough to measure system performance with ideal resolution. In order to better reflect the system performance, on the basis of the measured SNR, the above factors can be converted into quantization noise to get the ENOB. The calculation formula is as follows: ENOB is based on the equation for an ideal ADC's SNR: SNR = 6.02 × N + 1.76 dB, where N is the ADC's resolution.The difference between ENOB and ADC resolution reflects the degree of decrease in sampling accuracy caused by the decrease in SNR(here SNR caused by the error source). 5) Non-Linearity ErrorNon-linear error is an important accuracy index of the converter, which represents the difference between the actual conversion value of the ADC and the theoretical conversion value. Non-linear errors mainly include two types: Differential Non-Linearity (DNL) errors and Integral Non-Linearity (INL) errors. 6) Inter Modulation Distortion (IMD)When two sinusoidal signals are input to the ADC at the same time, due to the nonlinearity of the device, except the components of these two frequencies, the output spectrum will also produce many distortion products. The resulting distortion is called inter modulation distortion ( IMD, Inter Modulation Distortion), where the value of m+n represents the order of distortion. Among all inter-modulation distortions, the second-order and third-order inter-modulation products are the most important. The former is easily filtered out by a digital filter, while the latter is difficult to filter out. 7) Total Harmonic Distortion (THD)Due to the nonlinearity of the ADC, many high-order harmonics of the input signal appear in the output spectrum. These high-order harmonic components are called harmonic distortion components, and the resulting distortion is called Total Harmonic Distortion. Harmonic distortion and modulation distortion are two different concepts. The former is a distortion of the original signal waveform, even if a single frequency signal passes through the ADC, this phenomenon will occur, while the latter is mutual interference and influence between different frequencies.Figure 3. ADC on the ArduinoⅣ A/D Converter Applications and ICs4.1 Analog-to-Digital Converter ApplicationsMost ADC applications today belong to Four Segments: (a) Data acquisition(b) Precision industrial measurement(c) Voiceband and audio(d) High speed (sampling rates greater than about 5 MSPS)4.2 Analog-to-Digital Converter IC Modes ExplainedThere are many ADC ICs available in the market which can be used along to do conversion. Here lists several ADC ICs and their features and specifications as ADC selection references.⭕AD762116-Bit, 2 LSB INL, 3 MSPS PulSAR® ADC, High sampling rate, Available in a 48-lead LQFP or a 48-lead LFCSP⭕AD764118-Bit, 2 MSPS, Charge Redistribution SAR ADC 16 Bits Resolution with No Missing CodesNo Pipeline Delay ( SAR architecture )Differential Input Range: ±VREF (VREF up to 2.5V)Throughput:  3 MSPS (Wideband Warp and Warp Mode)  2 MSPS (Normal Mode)  1.25 MSPS (Impulse Mode)INL ±2 LSB Max (±30 ppm of FS)SINAD: 89 dB Typ @ 100 kHzTHD: -103 dB Typ @ 100 kHzParallel (16 or 8 bits bus) and Serial 5 V/3.3 V/2.5 V InterfaceSPI®/QSPI™/MICROWIRE™/DSP CompatibleOn-board Low Drift Reference with Buffer and Temperature SensorSingle 2.5 V Supply OperationPower Dissipation: 70 mW Typ @ 3 MSPS With REF18-bit resolution with no missing codes2.5 V internal low drift referenceThroughput:  2 MSPS (Warp mode)  1.5 MSPS (Normal mode)Differential input range: ± VREF (VREF up to 2.5 V)INL: ±2 LSB typicalNo pipeline delay (SAR architecture)Parallel (18-, 16-, or 8-bit bus)Serial 5 V/3.3 V/2.5 V interfaceSPI®/QSPI™/MICROWIRE™/DSP compatibleOn-board low drift reference with buffer and temperature sensor ⭕AD79088-Channel, 1 MSPS, 8-Bit ADC with Sequencer in 20-Lead TSSOP⭕AD79188-Channel, 1 MSPS, 10-Bit ADC with Sequencer in 20-Lead TSSOPFast throughput rate: 1 MSPSSpecified for AVDD of 2.7 V to 5.25 VLow Power:  6.0 mW max at 1 MSPS with 3 V supply  13.5 mW max at 1 MSPS with 5 V supplyEight (single-ended) inputs with sequencerWide input bandwidth: AD7928, 70 dB min SINAD at 50 kHz input frequencyFlexible power/serial clock speed managementNo pipeline delaysHigh speed serial interface SPI®/QSPI™/MICROWIRE™/DSP compatibleFast throughput rate: 1 MSPSSpecified for AVDD of 2.7 V to 5.25 VLow Power:  6.0 mW max at 1 MSPS with 3 V supply  13.5 mW max at 1 MSPS with 5 V supplyEight (single-ended) inputs with sequencerWide input bandwidth: AD7928, 70 dB min SINAD at 50 kHz input frequencyFlexible power/serial clock speed managementNo pipeline delaysHigh speed serial interface SPI®/QSPI™/MICROWIRE™/DSP compatible ⭕AD79288-Channel, 1 MSPS, 12-Bit ADC with Sequencer in 20-Lead TSSOP⭕AD5555Precision DUAL 16-Bit 14-Bit-DACs in Compact TSSOP PackagesFast throughput rate: 1 MSPSSpecified for AVDD of 2.7 V to 5.25 VLow Power:  6.0 mW max at 1 MSPS with 3 V supply  13.5 mW max at 1 MSPS with 5 V supplyEight (single-ended) inputs with sequencerWide input bandwidth: AD7928, 70 dB min SINAD at 50 kHz input frequencyFlexible power/serial clock speed managementNo pipeline delaysHigh speed serial interface SPI®/QSPI™/MICROWIRE™/DSP compatible14-bit resolution±1 LSB DNL monotonic±1 LSB INL2 mA full-scale current ±20%, with VREF = 10 V0.5 μs settling time2Q multiplying reference-input 6.9 MHz BWZero or midscale power-up presetZero or midscale dynamic reset3-wire interfaceCompact TSSOP-16 package ⭕AD823016 V Rail-to-Rail, Zero-Drift, Precision Instrumentation Amplifier⭕AD77993-Channel, Low Noise, Low Power, 24-Bit, Sigma Delta ADC with On-Chip In-AmpResistor programmable gain range: 101 to 1000Supply voltage range: ±4 V to ±8 VRail-to-rail input and outputMaintains performance over −40°C to +125°CExcellent ac and dc performance  110 dB minimum CMR @ 60 Hz, G = 10 to 1000  10 μV maximum offset voltage (RTI, ±5 V operation)  50 nV/°C maximum offset drift  20 ppm maximum gain nonlinearityRMS noise:  27 nV at 4.17 Hz (AD7799)  65 nV at 16.7 Hz (AD7799)  40 nV at 4.17 Hz (AD7798)  85 nV at 16.7 Hz (AD7798)Current: 380 μA typicalPower-down: 1 μA maximumLow noise, programmable gain, instrumentation ampUpdate rate: 4.17 Hz to 470 Hz 3 differential inputsInternal clock oscillatorSimultaneous 50 Hz/60 Hz rejectionReference detectLow-side power switchProgrammable digital outputsBurnout currentsPower supply: 2.7 V to 5.25 V ⭕AD944414-Bit, 80 MSPS A/D Converter⭕AD944514-Bit, 105 MSPS / 125 MSPS A/D Converter80 MSPS guaranteed sampling rate100 dB two-tone SFDR with 69.3 MHz and 70.3 MHz73.1 dB SNR with 70 MHz input97 dBc SFDR with 70 MHz inputExcellent linearity  DNL = ±0.4 LSB typical  INL = ±0.6 LSB typical1.2 W power dissipation3.3 V and 5 V supply operation2.0 V p-p differential full-scale inputLVDS outputs (ANSI-644 compatible)Data format selectOutput clock available125 MSPS guaranteed sampling rate (AD9445BSV-125)100 dB two-tone SFDR with 30 MHz and 31 MHz73.5 dB SNR with 70 MHz input85 dBc SFDR with 225 MHz inputExcellent linearity  DNL = ±0.25 LSB typical  INL = ±0.8 LSB typical2.3 W power dissipation3.3 V and 5 V supply operation2.0 V p-p to 3.2 V p-p differential full-scale inputLVDS outputs (ANSI-644 compatible) or CMOS outputsData format select (Offset Binary or 2’s compliment)Output clock available ⭕AD944616-Bit, 80 MSPS / 100 MSPS A/D Converter⭕AD923512-Bit, 20/40/65 MSPS, 3 V Analog-to-Digital Converter100 MSPS guaranteed sampling rate (AD9446-100)83.6 dBFS SNR with 30 MHz input(3.8 V p-p input, 80 MSPS)82.6 dBFS SNR with 30 MHz input(3.2 V p-p input, 80 MSPS)89 dBc SFDR with 30 MHz input(3.2 V p-p input, 80 MSPS)95 dBFS 2-tone SFDR with 9.8 MHz and10.8 MHz (100 MSPS)l 60 fsec rms jitterExcellent linearity  DNL = DNL = ±0.4 LSB typical  INL = ±3.0 LSB typical2.0 V p-p to 4.0 V p-p differential full-scale inputBuffered analog inputsLVDS outputs (ANSI-644 compatible) or CMOS outputsData format select (offset binary or twos complement)Output clock available3.3 V and 5 V supply operationSingle +3 V Supply Operation (2.7 V to 3.6 V)SNR = 70 dBc to Nyquist at 65 MSPSSFDR = 85 dBc to Nyquist at 65 MSPSLow Power: 300 mW at 65 MSPSOn-Chip Reference and SHADifferential Input with 500 MHz BandwidthDNL of ±0.4 LSBFlexible Analog Input: 1 V p-p to 2 V p-pOffset Binary or Twos Complement Data FormatClock Duty Cycle StabilizerPin out Migration to Either AD9215, AD9236, AD9245 Frequently Asked Questions about Analog to Digital Converter (ADC Basic)1. What is the use of analog to digital converter?Analog-to-digital converters, abbreviated as “ADCs,” work to convert analog (continuous, infinitely variable) signals to digital (discrete-time, discrete-amplitude) signals. In more practical terms, an ADC converts an analog input, such as a microphone collecting sound, into a digital signal. 2. What are the types of analog to digital converters?There are really five major types of ADCs in use today:Successive Approximation (SAR) ADCDelta-sigma (ΔΣ) ADCDual Slope ADCPipelined ADCFlash ADC 3. Which chip is used in analog to digital?An A/D converter is used to convert an analog signal like voltage to digital form so that it can be read and processed by a microcontroller. Some microcontrollers have built-in A/D converters. It is also possible to connect an external A/D converter to any type of microcontroller. 4. Which circuit is used in analog to digital converter?Analog to Digital Converter (ADC) is an electronic integrated circuit used to convert the analog signals such as voltages to digital or binary form consisting of 1s and 0s. Most of the ADCs take a voltage input as 0 to 10V, -5V to +5V, etc., and correspondingly produces digital output as some sort of a binary number.