【音频测试基础知识】加权滤波器

A, B, C and D weighting


The A-weighting curve is one of a set of four, defined in various standards relating to sound level measurement as A, B, C, and D. Curves A, B and C are for low, medium and high loudness sounds. D is specifically for measuring very loud aircraft noise.

The A-weighing curve is based on the 40-phon equal-loudness contour for typical human hearing, being a rough approximation of that curve after inversion to indicate gain rather than level. Although our absolute threshold of hearing is around 0 Phons, this is a very quiet level indeed, and only realised in specially isolated conditions. 40 phon is the sort of level likely to exist in a normal quiet environment.

While early experimenters had to approximate roughly in order that the A-weighing curve could be realised economically in a filter that did not need too many components, today we could easily realise the true curve. This has not been done, but a revision is arguably overdue, especially as the A-weighting curve is not defined above 20kHz and fails to reflect the steep cut-off in our hearing above 15kHz which is also not made clear by the equal loudness contours, which aim to quantify only our in-band response. Unless the A-weighting filter is used in conjunction with a band-limiting filter cutting off above 20kHz, it cannot be used with any validity for loudness measurement, since ultrasonic noise from many sources can affect the measurement – a fact that is usually overlooked. More damning though, is the fact that equal loudness contours are now known to be invalid for noise measurement, even though this what sound level meters incorporating A-weighting are mostly used for! The later ITU-R weighting curve should therefore be considered for all noise measurement purposes, though this is currently only used in broadcasting and professional audio, and is not usually incorporated into sound level meters.

2 Equal-loudness contours

In 1933 Fletcher and Munson (1933) investigated the variation in apparent loudness of sounds with frequency and found that they needed to plot a set of 'equal loudness' contours, because the ear’s frequency response varied with loudness level. They presented volunteer subjects with pure tones at various frequencies, and asked them to adjust the level until they judged each tone to sound equal in loudness to a 1kHz reference tone. Then they repeated the experiment with different levels of reference tone. This revealed that our ears are much less sensitive to low frequency sounds, especially at low levels. Robinson and Dadson repeated the experiment in 1956, and obtained somewhat different results, which were considered to be more accurate, and became the basis for the ISO standard, ISO226 until modern experiments began to cast doubt on their validity, resulting in a survey by the International Standards Organisation. This led to the release, in 2003, of a new standard: ISO 226:2003, which was derived from the reported results of various experimenters. The graphs shown here compare first the Fletcher-Munson and then the Robinson-Dadson results with the modern curves, and it can be seen that while the former appear to be badly out at high levels in the low-frequency region they are in quite good agreement at low levels. The Robinson-Dadson curves, which until 2003 formed the basis for ISO226, differ significantly, showing around 12dB greater sensitivity in the region around 150Hz. The recent ISO survey report commented that it was fortunate that the 40-phon contour of Fletcher and Munson, on which the A-weighting filter was based, does agree quite well with the new ISO226 standard.

3 Possible errors

The reasons for these discrepancies appear not to have been investigated, but several factors make the experiment very difficult. Firstly, loudness is only a subjective quality, and it is quite hard to say when two tones at different frequencies sound equally loud. This is especially true at frequencies below 100Hz where we sense the sound less as a tone and more as a feeling.

Then there is the difficulty of providing an undistorted sound source of known level. Even today, most loudspeakers have at least one percent of harmonic distortion at most frequencies, and very few indeed would be able to achieve even this at 20Hz and 100dB SPL. Fletcher and Munson used headphones, but in 1933 most headphones had poor low-frequency response, and it seems likely that what their subjects were hearing below 100Hz consisted mostly of higher harmonics, which would explain their apparently superior sensitivity at high levels. The steep slope of the curves at low frequencies means of course that we are very sensitive to harmonics, and it is now well known that our brains even have the special ability to ‘postulate’ a missing fundamental, so that we think we hear the fundamental even when it is not there, a phenomenon that makes small loudspeakers sound more acceptable than they otherwise would!

Robinson and Dadson used loudspeakers for their experiment, and one can only wonder at how they managed to generate a flat distortion-free level of 125dB SPL at 20Hz in 1956 (if they did)! Their results show an almost level response from 1kHz down to 200Hz compared to a 10dB loss of sensitivity on the modern ISO 226 curve, in a region that poses no special difficulties for loudspeaders. It is possible that a room resonance or reinforcing reflection off the floor caused the apparent increase in sensitivity around 150Hz, because today a large room surrounded by absorbent wedges on all sides, together with a false floor supporting the listener, would be necessary to avoid such effects. However, one would expect them to have been fully aware of these problems, which would also be expected to cause anomalies at lower frequencies.

To confuse matters, my own tests using headphones, carefully calibrated using an in-ear microphone, on my own ears, show no fall in sensitivity at 200Hz! Were Robinson and Dadson right after all, and if so why the discrepancy in the new ISO 226 curve? One explanation I can suggest is that the ear’s low frequency response depends on the tautness of the eardrum, which is governed by the stapedius and tensor timpani muscles, which tense under loud conditions, and remain tensed for hours and even days. Several modern studies were by Japanese researchers. Could it be that their subjects were adapted to noisy city life, and not kept in quiet conditions for several days prior to and during testing?

Another problem that arises when trying to generate very high levels at low frequencies for these experiments is the need for an extremely low level of amplifier noise, because any hiss is not masked by the tone, and so must be kept 125dB or more below the tone if it is to be inaudible. This is just about impossible today, and inconceivable in the days of valves (vacuum tubes), though a possible way round the problem is to use an acoustic filter during the low-frequency tests (such as a quilt hung in front of the loudspeaker!).

4 Frontal versus side presentation

A complicating factor in the derivation of equal-loudness contours is the fact that our ears have a different frequency response in every direction. This becomes increasingly true above about 2kHz, and does not affect low frequencies, being caused by head-masking and the complicated shape of the outer ear, or pinna.

Very low frequencies pass around the head and are sensed equally by each ear purely as pressure variation, but as the wavelength of a sound becomes comparable with the dimensions of our head (1kHz equates to 320mm or roughly 1 foot) this starts to get in the way so that each ear becomes progressively less sensitive to sound from its opposite side as frequency increases. At still higher frequencies the outer ear begins to focus the sound, forming a cavity with various resonances that are excited to varying degrees by sounds from different directions, in both the horizontal and vertical planes. These two mechanisms, both involving difference in loudness between the ears are used by the brain to enable us to locate the direction of sounds in both the horizontal and vertical planes, along with another mechanism which uses the different time-of arrival at each ear, or phase difference, to determine direction. The latter mechanism takes over in the lower 100Hz to 2kHz region, and is made use of in stereo reproduction, though it cannot give us information about sounds originating above or behind us where the pinna comes into effect.

Head masking and pinna effects have been quantified in terms of another set of contours referred to as ‘Head-related transfer functions’, which also vary considerably between individuals at high frequencies. They are relevant to the determination of equal-loudness contours because we are considerably more sensitive to high frequency sound coming directly from each side, as in headphone listening, than to sound from a single loudspeaker at centre-front, with stereo presentation from two speakers at 60 degrees separation coming somewhere in between. Modern equal-loudness contours are for frontal presentation, but although Fletcher and Munson used headphones, the ISO survey lists their experiment as using ‘compensated headphones’. Most modern headphones intended for listening to stereo are in fact designed with a severe dip in their response around 3-10kHz, without which they would sound too bright; a fact that is not commonly recognised.
5 Towards a better noise-weighting standard

A modern version of A-weighting might be expected to take the form of the inverse 40-phon ISO 226 curve, as shown here compared to both the traditional A-weighting curve and the later ITU-R 468 weighting. This is not necessarily the best shape for it though, because above 2kHz each ISO curve is the result of averaging results from many individuals, some of whom have dips where others have peaks. A more sensible approach would therefore be to plot the results for many individuals and then draw a smooth curve through the maxima, since for noise measurement purposes we are more interested in what someone might hear than in what one person does not hear. We might then choose to err on the high side (by perhaps 6dB) around 6kHz, because it is now known that we are most sensitive to random noise in this region, as reflected in the ITU-R 468 weighting curve, and in practice environmental noise tends to be part tonal (music and some machinery) and part random (wind noise, traffic, and machinery).

Sound level meters currently use rms detection, which would be reasonable if our perception of loudness was related to power, but it in not. A series of clicks or tone-bursts sound almost as loud as the continuous sound, provided that the gaps between them are not too long, showing that our perception of loudness has more to do with peak level than average or rms. As the duration of a tone-burst is reduced below about 50ms, however, is starts to sound progressively quieter, showing that our ears take time to respond in each frequency band. To be perceptually valid therefore, sound level and noise level measurements must use some form of ‘quasi-peak’ detector with carefully devised integration times, as is the case with the ITU-R 468 standard. The traditional sound-level meter with traditional A-weighting is therefore starting to look rather irrelevant for all practical purposes.

6 Computed noise-measurement

Better still would be a system that aimed to model our hearing characteristics using a bank of overlapping narrow-band filters, with varying bandwidths, spaced across the audible spectrum. Their outputs would be combined using an algorithm that also incorporated quasi-peak detection and non-linearity to produce a result that remained reasonably valid for all types of sound at all levels. This need not be too difficult using modern digital processing, and would be relevant for all applications, from environmental and aircraft noise measurement to measurements on audio systems and telephone networks.

o-o-o-o-o-o-o-o-o-o-o-o-o-o

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【音频测试基础知识】PMPO音乐功率真相

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【音频测试基础知识】信噪比与动态范围

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【音频测试基础知识】噪音测量

Introduction to noise measurement

Noise in general refers to unwanted sound, often loud, but in audio systems it is the low-level hiss or buzz that intrudes on quiet passages that is of most interest. All recordings will contain some background noise that was picked up by microphones, such as the rumble of air conditioning, or the shuffling of an audience, but in addition to this every piece of equipment which the recorded signal subsequently passes through will add a certain amount of electronic noise, which ideally should be so low as to contribute insignificantly to what is heard.

Microphones, amplifers and recording systems all add some electronic noise to the signals passing through them, generally described as hum, buzz or hiss. All buildings have low-level magnetic and electrostatic fields in and around them emanating from mains supply wiring, and these can induce hum, commonly at 50 or 100Hz into signal paths. Screened cables help to prevent this, and on professional equipment, where longer interconnections are common, balanced signal connections (most often with XLR connectors) are usually employed. Hiss is the result of random signals, often arising from the random motion of electrons in transistors and other electronic components, or the random distribution of oxide particles on analog magnetic tape. It is predominantly heard at high frequencies, sounding like steam or compressed air.

Attempts to measure noise in audio equipment as rms voltage, using a simple level meter or voltmeter, do not produce useful results; a special noise-measuring instrument is required. This is because noise contains energy spread over a wide range of frequencies and levels, and different sources of noise have different spectral content. For measurements to to allow fair comparison of different systems they must be made using a measuring instrument that responds in a way that corresponds to how we hear sounds. From this, three requirements follow. Firstly, it is important that frequencies above or below those that can be heard by even the best ears are filtered out and ignored; by bandwidth limiting (usually 22Hz to 22kHz). Secondly, the measuring instrument should give varying emphasis to different frequency components of the noise, in the same way that our ears do; a process referred to as ‘weighting’. Thirdly, the rectifier, or detector, which is used to convert the varying alternating noise signal into a steady positive representation of level should take time to respond fully to brief peaks to the same extent that our ears do; it should have the correct ‘dynamics’.

The proper measurement of noise therefore requires the use of a specified method, with defined measurement bandwidth and weighting curve, and rectifier dynamics, and two main methods defined by standards are currently in common use: A-weighing, and ITU-R 468, formerly known as CCIR weighting.

A-weighting

A-weighting uses a weighting curve based on ‘equal loudness contours’ that describe our hearing sensitivity to pure tones, but it turns out that the assumption that such contours would be valid for noise components was wrong. While the A-weighting curve peaks by about 2dB around 2kHz, it turns out that our sensitivity to noise peaks by some 12dB at 6kHz. Another weakness of A-weighting is that it is usually combined with an rms (root mean square) rectifier, which measures mean power, with no attempt made to account for proper hearing dynamics.

ITU-R 468 weighting

When measurements started to be used in reviews of consumer equipment in the late 1960’s it became apparent that they did not always correlate with what was heard. In particular, the introduction of Dolby B noise-reduction on cassette recorders was found to make them sound a full 10dB less noisy, yet they did not measure 10d better. Various new methods were then devised, including one which used a harsher weighting filter and a quasi-peak rectifier, defined as part of the German DIN45 500 ‘Hi Fi’ standard. This standard, no longer in use, attempted to lay down minimum performance requirements in all areas for ‘High Fidelity’ reproduction.

The introduction of FM radio, which also generates predominantly high-frequency hiss, also showed up the unsatisfactory nature of A-weighting, and the BBC Research Department undertook a research project to determine which of several weighting filter and rectifier characteristics gave results that were most in line with the judgment of panel of listeners, using a wide variety of different types of noise. BBC Research Department Report EL-17 formed the basis of what became known as CCIR recommendation 468, which specified both a new weighting curve and a quasi-peak rectifier. This became the standard of choice for broadcasters worldwide, and it was also adopted by Dolby, for measurements on its noise-reduction systems which were rapidly becoming the standard in cinema sound, as well as in recording studios and the home.

Though they represent what we truly hear, CCIR weighted noise figures are typically some 11dB worse than A-weighted, a fact that brought resistance from marketing departments reluctant to put worse specifications on their equipment than the public had been used to. Dolby tried to get round this by introducing a version of their own called CCIR-Dolby which incorporated a 6dB shift into the result (and a cheaper average reading rectifier), but this only confused matters, and was very much disapproved of by the CCIR.

With the demise of the CCIR, the 468 standard is now maintained as ITU-R 468, by the International Telecommunications Union, and forms part of many national and international standards, in particular by the IEC (International Electrotechnical commission), and the BSI (British Standards Institute). It is by far the best way to measure noise, and the only way that allows fair comparisons; and yet the flawed A-weighting has made a comeback in the consumer field recently, for the simple reason that it gives the lower figures that are considered more impressive by marketing departments.

Signal to noise ratio and Dynamic range

Hi-fi equipment specifications tend to include the terms ‘signal to noise ratio’ and ‘dynamic range’, both of which are confusing and best avoided. Noise has to be measured with reference to something, but this should be ‘alignment level’. Signal to noise ratio has no real meaning as audio signals are constantly changing so there is no such thing as ‘signal level’. Dynamic range used to mean the difference between maximum level and noise level, but maximum level is often hard to define, for example on analog tape recordings, and the term has become corrupted by a tendency to refer to the dynamic range of CD players as meaning the noise level on a blank recording with no dither, in other words just the analog noise content at the output. This is not particularly useful; especially since many CD players incorporate automatic muting in the absence of signal to make them appear even quieter!

Professionals measure noise in dB below alignment level, which is a reference point above which ‘headroom’ exists up to maximum permitted level. Professionals often allow 18dB of headroom, as recommended by the EBU (European Broadcasting Union), so a noise level of –60dB ITU-R 468 would represent a dynamic range of 78dB, which if measured A-weighted might come out 11dB better at 89dB. A noise level of -60dB AL would be considered reasonably good by professionals, with –68dB representing the best attainable from 16-bit digital audio (noise shaped), and more than good enough for most purposes.

Audiophiles may talk in terms of 96 to 120dB dynamic range, but they often fail to refer to any measurement standard, making the figures meaningless. Attempts to calculate the dynamic range of digital audio on the basis that 16 bits represents a ratio of 65000:1 or 96dB are invalidated by the fact that the full digital count represents the peak possible level, rather than the rms equivalent of the maximum possible sinewave, while the minimum count of one has little to do with the noise level, which depends on the type of dither (or noise-shaping) used. They also fail to take any account of weighting for subjective validity.

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【音频测试基础知识】串音测量

Crosstalk measurement is made on audio systems to determine the amount of signal leaking across from one channel to another.

Interchannel crosstalk between the two channels of a stereo system, and is usually not very important on modern systems, though it was hard to keep below the desired figure of -30dB or so on vinyl recordings and FM radio.

A relatively high level of crosstalk between stereo channels is normally of no consequence, since most sound sources will have been mixed to both channels to some extent anyway, and a small extra amount due to crosstalk will not normally affect the stereo image. It is important though that any significant crosstalk is not in the form of distortion, something that is often missed in relation to power amplifiers. In a power amplifier the currents flowing in power rails can be essentially half-wave rectified signals if they original from one side of a push-pull output circuit. If these waveforms couple into the other channel they can contribute serious distortion. Though not a common problem, this is a possibility that should be borne in mind.

Crosstalk between channels in mixing consoles, and between studio feeds is much more of a problem, as these are likely to be carrying very different programmes or material which is likely to be heard in quiet passages unless it is at a very low level.

The IBA drew up a weighting curve for use in crosstalk measurement that gives due emphasis to the subjective audibility of different frequencies, as shown here. This is still in use, despite the demise of the IBA, and in the absence of any international standards is worth adopting.

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【音频测试基础知识】失真测量

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理解dB

Understanding decibels (dB)

The decibel is widely used as a measure of the loudness of sound, but it is actually only a convenient way of specifying the ratio between two quantities, so a 6' 7" man could be said to be 1dB taller than a 6' man! An engineer might use it for such things in jest, but in general the dB is commonly used to express a wide variety of measurements in acoustics and electronics, where it helps in giving a manageable view of things that can cover huge range of values, where only geometric increase or decrease is important. It is especially useful in audio because our ears follow a roughly logarithmic law, referred to by psychologists as Weber's law. The smallest change in loudness that we can detect is about 1dB, or a ten percent change, regardless of whether the sound is very quiet or very loud, in other words we hear loudness 'geometrically' rather than 'arithmetically'.

The decibel is a dimensionless unit like percent and only has meaning when levels are specified relative to a reference level. In audio, the reference level is often specified with a suffix after the dB notation and a brief summary of the most commonly used standards follows:

dBm

dBm specifies a power level on a line, referenced to 1mW. In the early days of audio signals were passed along 'matched' transmission lines, meaning that both the source impedance and the load impedance were the same as the 'characteristic impedance' of the line, usually 600 ohms. This ensured that all energy sent down the line ended up in the load resistance, with non reflected back to cause frequency response anomalies or echoes.

dBu

dBu specifies that the voltage amplitude of an audio signal is referenced to 0.775 volts rms. This is the same voltage as would be needed to dissipate 1mW into a 600 ohm resistor, and is kept for historical reasons, though 600 ohm matched transmission is rarely used today. The standard for studio interconnection today uses low source impedance (<50 ohms) and high input impedance (10 or 20k ohm). The use of a 'solid' drive ensures more accurate levels, and reflections are not a problem over short paths (say <50m). The u is thought to stand for 'universal' or maybe 'unloaded'. 0dBu is the universal 'Alignment level' within many broadcast organisations and recording studios, with signal levels allowed into the 'headroom' region which may be 24dB, 18dB (EBU recommendation for programme interchange) or 8dB (EBU standard for radio and TV broadcasts and paths).
dBV

dBV specifies that the reference amplitude is 1V. This is sometimes used for consumer levels, but dBu are to be preferred because most audio instruments are calibrated in dBu.


dB SPL

This is a measure of sound pressure level, relative to 20 micropascals (µPa = 1×10^-6 Pa), an arbitrary figure chosen as being about the quietest sound a human can hear. This is roughly the sound of a mosquito flying 3 metres away!

dB FS

This means relative to full-scale or the point at which clipping occurs in a system. It is frequently used in referring to digital signal levels which do not in themselves correspond to any particular voltage, until they are converted using a D-A convertor.

dB AL (Recommended by Lindos)

This means dB relative to Alignment Level, which is a reference level or anchor point against which all else in an audio system is measured. Alignment level may correspond to different voltage levels in various part of the system or signal path (though commonly it will be 0dBu), and in a digital recording it may correspond with varying digital values, though commonly it will be -18dB FS (or -24 or -12) as defined in EBU recommendations. Importantly it is the reference at any point in a system above which Headroom is defined and relative to which noise is measured.

The great merit of this system is that a noise measurement made relative to Alignment Level will always be a reasonable guide to the true intrusiveness of the noise onto typical material, regardless of the headroom. Where less headroom is available for transmission, compression or limiting will usually be used to reduce brief peaks, but this has little effect on overall perceived loudness. A recording may start life with 24dB of headroom, and end up with only 9dB of headroom at the listener’s radio or television. Its ‘dynamic range’ has indeed been reduced, but it will not sound any noisier, assuming that it is the peaks that have been limited. Instead it will have lost ‘sparkle’ and impact. Specifying noise and headroom separately reflects these different qualities properly.

Decibels in Audio Recording and Reproduction

The dynamic range of the human ear is phenomenal, through a complex gain adjustment system a range of around 140dB is covered. Accurately capturing and reproducing the quietest and loudest sounds audible by humans is a formidable task.

The diagram below shows the range of levels handled by various stages in a typical audio chain, from live sound to loudspeaker. Note how most devices cannot cope with the full range without the use of compression or limiting.

 


Useful Conversion Tables

It is useful to remember that 6dB is approximately a factor of 2.0, 10dB a factor of 3, and 20 dB exactly 10 times, when referring to voltage levels. When levels are multiplied, dB are added. Thus 26dB represents 20 times.

Power levels, being proportional to the square of the voltage, have different ratios, so that 3dB is twice the power, 6dB is 4 times the power, and 10dB is ten times the power. Usually, these days, it is voltage ratios that are relevant, and the following table will be found useful to commit to memory:


0dB x 1
+1dB x 1.1
0dB x 1
+1dB x 1.1
+3dB x 1.414 (root 2)
+6dB x 2
+10dB x 3
+20dB x 10
+30dB x 30
+40dB x 100
+50dB x 300
+60dB x 1000 etc

and:

-1dB x 0.9
-3dB x 0.707
-6dB x 0.5
-10dB x 0.3
-20dB x 0.10
-40dB x 0.01
-60dB x 0.001 etc
o-o-o-o-o-o-o-o-o-o

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气压表做好了

大气压力表做好了!

不仅能测大气压力，而且还能测温度，查大气压力和海拔对应数据表，还能显示海拔值。传感器的精度大气压力0.01hpa，温度0.1度，海拔0.1m

这是做好PCB（第一版，还不是正式产品），板子上做好了RS485和RS232口，方便与其它设备相连。

配上原来做好的USB板，还能增加USB端口。用的是CH375

连上电脑，看看数据如何？

这是串口出来的数据：

用Labview编了个仪器面板，看起来专业多了：

看看现在仪表上的数值，大概能猜到我在什么地理位置了吧，对，是在又热又闷的海边（海拔才几米）

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分析一个防静电手腕带测试仪电路

有线静电手腕带是防止静电的有效工具，为了保证手腕带正常工作，工厂生产线的QA经常会用一种手腕带试仪来测试员工佩戴的手腕带的好坏。

静电手腕带的构造很简单，一条弹性导电布带，再加上一条接地线，接地线和腕带之间串有一1MΩ的电阻。

手腕带测试仪用来测试手腕带是否出现接地线脱落、腕带与皮肤接触不良等（工厂经常有员工把手腕带套在衣服袖子上），其原理很简单，就是测试人体到地的电阻，太大、太小都是不良。

工厂的测试仪坏了，正好趁着维修之机来看看它是怎么工作的。电路图用手描的，来不及上电脑了。

手环的测试方法就像图片中的一样，接地线接图纸左上角的那个接地点，然后带着手环按下金属的测试按键，人体接触到第二个触点，同时按下了开关。

电路很简单，一片LM324，一片4558. 单电源工作。

人体+手环+接地线的电阻的加入，引起1MΩ电阻上的电压发生变化，,三个运放构成三个电压比较器，分别检测电压变化是否超出上下限、是否在OK范围内，同时分别点亮三颗LED和蜂鸣器。

4558完成成电池电量低指示功能。

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用PROTEUS仿真研究直流马达特性

在学习电机控制之前，当然要对控制对象-直流电机的特性要有所了解才行。严格说来电机本身就是一个相当复杂的对象，再加上传动系统更是复杂，而且普通人缺乏必要的仪器设备，难以对电机进行深入了解。幸运的是我们有Proteus这样优秀的仿真软件，在它的元件库中可以找到许多种类型的马达元件，从简单的直流马达到步进马达、伺服马达都有，为我们进行仿真研究提供了非常好的条件。

电机控制最重要的课题就是速度和位置的控制，我们先来研究一下直流马达的基本特性，这对掌握电机的控制方法是相当重要的。

马达一上电就开始旋转，转速从慢到高，直到转速不能再提高为止，这个过程在实际应用当中是经常见到的，但是马达的转速究竟是怎么变化的，和驱动电压、电机本身的特性、负载又有什么关系？这是需要研究的.....

在Proteus中驱动一个马达是很容易的，我们选取一个带编码器的马达来做试验，之所以用带编码器的，是因为我们需要用到编码器的输出信号来测量马达的转速。

编码器输出信号的详细说明在Help文件中可以找到。

双击马达元件，进入参数设置界面，把其中"Pulse Per Revolution"一项设置成60,即马达每旋转一周，编码器输出60个脉冲，这样做的好处下面就可以看到。

给Drive端子加高电平，马达就可以慢慢旋转起来，马达下面的绿色数字就是马达的转速，单位是rpm即"转/分钟",正负号表示顺时针转还是逆时针转。

这时候用一个频率计测量编码器输出脉冲，可以看到频率值正好和转速相同，因为我们设置了每转60个脉冲，频率值正好与每分钟的转数相当。

要了解马达的特性，这还不够，因为我们只看到了转速的快速变化却没法精确记录每时刻的速度值。在Proteus中也没有这种记录功能（谁找到了请告诉我）。这时候，单片机登场了…..

思路是这样的：用单片机做一个频率计，然后把每一秒钟的转速值记录下来（很容易，是吧？）然后通过串口把这些数据发出来! 我们得到这些数据就可以保存在数据文件中，进行事后分析了。这个方法听起来不错吧!

当然，更好的方法是用虚拟终端，因为很容易把数据Copy出来。

COPY之后，用编辑器新建一个文件，粘贴，然后保存为后缀为.csv的文件，用.csv后缀是因为我们要用Office EXCEL来处理数据

我这里做了一个简单的试验，看看马达的负载对马达加速的影响。首先把马达的负载率设为5%，然后加驱动电压，记录马达启动后每一秒的转速（也就是编码器输出脉冲的频率），共记录了100秒；之后，分别再设负载率为10%，20%，50%，做同样的记录。

整理所有数据到EXCEL中

做图表进行比较：

有兴趣的朋友可以和我一起研究学习，

电路图和程序在这：

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