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Amplifer Classes rom A to H
Engineers and audiophiles have one thing in common when it comes to amplifers. They want a design that
provides a strong balance between perormance, eciency, and cost.
I you are an engineer interested in choosing or designing the amplifer best suited to your needs, you’ll fnd
columnist Robert acoste’s article in Circuit Cellar ’s !ecember issue helpul. "is article provides a
comprehensive loo# at the characteristics, strengths, and wea#nesses o di$erent amplifer classes so you
can select the best one or your application.
The article, logically enough, proceeds rom %lass & through %lass " 'but only touches on the more nebulous
%lass T, which appears to be a developer’s custom(made creation).
*Theory is easy, but diculties arise when you actually want to design a real(world amplifer,+ acoste says.
*hat are your particular choices or its fnal ampliying stage-+
The ollowing article ecerpts, in part, answer that /uestion. '0or uller guidance, download Circuit
Cellar ’s !ecember issue.)
CLASS A
The frst and simplest solution would be to use a single transistor in linear mode (see Figure 1 )… Basically
the transistor must be biased to have a collector voltage close to V CC /2 when no signal is applied on the
input This enables the output signal to swing
Figure 1—A Class-A amplifier can be built around a simple transistor. The transistor must be biased in so it stays in the
linear operating region (i.e., the transistor is always conducting.
either above or below this !uiescent voltage depending on the input voltage polarity…
This solution"s advantages are numerous# simplicity$ no need %or a bipolar power supply$ and e&cellent
linearity as long as the output voltage doesn"t come too close to the power rails This solution is considered
as the per%ect re%erence %or audio applications But there is a serious downside
Because a continuous current 'ows through its collector$ even without an input signal"s presence$ this
implies poor eciency n %act$ a basic Class*+ amplifer"s eciency is barely more than ,-.…
CLASS B
ow can you improve an amplifer"s eciency0 1ou want to avoid a continuous current 'owing in the output
transistors as much as possible
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Class*B amplifers use a pair o% complementary transistors in a push*pull confguration (see Figure 2 ) The
transistors are biased in such a way that one o% the transistors conducts when the input signal is positive and
the other conducts when it is negative Both transistors never conduct at the same time$ so there are very
%ew losses The current always goes to the load…
+ Class*B amplifer has more improved eciency compared to a Class*+ amplifer This is great$ but there is
a downside$ right0 The answer is un%ortunately yes
The downside is called crossover distortion…
Figure !—Class-" amplifiers are usually built around a pair of complementary transistors (at left. #ach transistor conducts
$%& of the time. This minimi'es power losses, but at the epense of the crosso)er distortion at each 'ero crossing.
CLASS AB
+s its name indicates$ Class*+B amplifers are midway between Class + and Class B ave a loo at the
Class*B schematic shown in Figure 2. % you slightly change the transistor"s biasing$ it will enable a small
current to continuously 'ow through the transistors when no input is present This current is not as high as
what"s needed %or a Class*+ amplifer owever$ this current would ensure that there will be a small overall
current$ around 3ero crossing
4nly one transistor conducts when the input signal has a high enough voltage (positive or negative)$ but
both will conduct around - V There%ore$ a Class*+B amplifer"s eciency is better than a Class*+ amplifer
but worse than a Class*B amplifer 5oreover$ a Class*+B amplifer"s linearity is better than a Class*B
amplifer but not as good as a Class*+ amplifer
These characteristics mae Class*+B amplifers a good choice %or most low*cost designs…
CLASS C
There isn"t any Class*C audio amplifer 6hy0 This is because a Class*C amplifer is highly nonlinear ow can
it be o% any use0 +n 78 signal is composed o% a high*%re!uency carrier with some modulation The resulting signal is o%ten
!uite narrow in terms o% %re!uency range 5oreover$ a large class o% 78 modulations doesn"t modi%y the
carrier signal"s amplitude
8or e&le$ with a %re!uency or a phase modulation$ the carrier pea*to*pea voltage is always stable n
such a case$ it is possible to use a nonlinear amplifer and a simple band*pass flter to recover the signal9…
+ Class*C amplifer can have good eciency as there are no lossy resistors anywhere t goes up to :-. or
even ;-.$ which is good %or high*%re!uency designs 5oreover$ only one transistor is re!uired$ which is a ey
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CLASS G AND CLASS H
Class D and Class are !uests %or improved eciency over the classic Class*+B amplifer Both wor on the
power supply section The idea is simple 8or high*output power$ a high*voltage power supply is needed 8or
low*power$ this high voltage implies higher losses in the output stage
6hat about reducing the supply voltage when the re!uired output power is low enough0 This scheme is
clever$ especially %or audio applications 5ost o% the time$ music re!uires only a couple o% watts even i% %ar
more power is needed during the %ortissimo agree this may not be the case %or some teenagers" music$ but this is the concept
Class D achieves this improvement by using more than one stable power rail$ usually two Figure 4shows
you the concept
Figure 0—A Class- amplifier uses two pairs of power supply rails. b—2ne supply rail is used when the output signal has a
low power (blue. The other supply rail enters into action for high powers (red. +istortion could appear at the crosso)er.
What is Switch-mode power supply? The electronic power supply integrated with the switching regulator or converting
the electrical power eciently rom one orm to another orm with desired
characteristics is called as 1witch(mode power supply. It is used to obtain regulated
!% output voltage rom unregulated &% or !% input voltage.
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Switch mode power supply
1imilar to other power supplies, switch(mode power supply is a complicated circuit
that supplies power rom a source to loads. switch(mode power supply is essential
or power consuming electrical and electronic appliances and even or
building electrical and electronic pro2ects.
Topologies of Switch Mode Power Supply
There are di$erent types o topologies or 1341, among those, a ew are as ollows
• !% to !% converter
• &% to !% converter
• 0ly bac# converter
• 0orward converter
Switch Mode Power Supply’s Working Principle
The wor#ing o a ew types o switch(mode power supply topologies is as ollows5
1. ! to ! !on"erter SMPS Working Principle
In a !%(to(!% converter, primarily a high(voltage !% power is directly obtained rom
a !% power source. Then, this high(voltage !% power is switched at a very high
switching speed usually in the range o 67 8"9 to 7: 8"9.
&nd then it is ed to a step(down transormer which is comparable to the weight and
si9e characteristics o a transormer unit o 7:"9. The output o the step(down
transormer is urther ed into the rectifer. This fltered and rectifed output !%
power is used as a source or loads, and a sample o this output power is used as a
eedbac# or controlling the output voltage. ith this eedbac# voltage, the ;< time
o the oscillator is controlled, and a closed(loop regulator is ormed.
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DC to DC converter SMPS
The output o the switching(power supply is regulated by using 43 '4ulse idth
3odulation). &s shown in the circuit above, the switch is driven by the 43
oscillator, such that the power ed to the step(down transormer is controlled
indirectly, and hence, the output is controlled by the 43, as this pulse width signal
and the output voltage are inversely proportional to each other.
I the duty cycle is 7:=, then the maimum amount o power is transerred through
the step(down transormer, and, i duty cycle decreases, then the amount o power
transerred will decrease by decreasing the power dissipation.
#. $! to ! !on"erter SMPS Working Principle
The &% to !% converter 1341 has an &% input. It is converted into !% by
rectifcation process using a rectifer and flter. This unregulated !% voltage is ed to
the large(flter capacitor or 40% '4ower 0actor %orrection) circuits or correction o
power actor as it is a$ected. This is because around voltage pea#s, the rectifer
draws short current pulses having signifcantly high(re/uency energy which a$ects
the power actor to reduce.
AC to DC converter SMPS
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It is almost similar to the above discussed !% to !% converter, but instead o direct
!% power supply, here &% input is used. 1o, the combination o the rectifer and
flter, shown in the bloc# diagram is used or converting the &% into !% and
switching is done by using a power 3;10ET amplifer with which very high gain can
be achieved. The 3;10ET transistor has low on(resistance and can withstand high
currents. The switching re/uency is chosen such that it must be #ept inaudible to
normal human beings 'mostly above >:8"9) and switching action is controlled by a
eedbac# utili9ing the 43 oscillator.
This &% voltage is again ed to the output transormer shown in the fgure to step
down or step up the voltage levels. Then, the output o this transormer is rectifed
and smoothed by using the output rectifer and flter. & eedbac# circuit is used to
control the output voltage by comparing it with the reerence voltage.
%. &ly-'ack !on"erter type SMPS Working Principle
The 1341 circuit with very low output power o less than 6:: 'watts) is usually o
0ly(bac# converter type 1341, and it is very simple and low( cost circuit compared
to other 1341 circuits. "ence, it is re/uently used or low(power applications.
Fly-bac Converter type SMPS
The unregulated input voltage with a constant magnitude is converted into a
desired output voltage by ast switching using a 3;10ET? the switching re/uency is
around 6:: #"9. The isolation o voltage can be achieved by using a transormer.
The switch operation can be controlled by using a 43 control while implementing
a practical @y(bac# converter.
0ly(bac# transormer ehibits di$erent characteristics compared to general
transormer. The two windings o the @y(bac# transormer act as magnetically
coupled inductors. The output o this transormer is passed through a diode and a
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capacitor or rectifcation and fltering. &s shown in the fgure, the voltage across
this flter capacitor is ta#en as the output voltage o the 1341.
(. &orward !on"erter type SMPS Working
0orward converter type 1341 is almost similar to the 0ly(bac# converter type 1341,but in the orward converter type, a control is connected or controlling the switch
and at the output o the secondary winding o the transormer, and the rectifcation
and fltering circuit is complicated as compared to the @y(bac# converter.
It can be called as a !% to !% buc# converter, along with a transormer used or
isolation and scaling. In addition to the diode !6 and capacitor %, a diode !> and an
inductor are connected at the output end. I switch 1 gets switched ;. Ay using the flter inductor, the re/uired
voltage across the diode !> and to maintain the E30 re/uired or maintaining the
continuity o the current at inductive flter.
Even though the current is diminishing against the output voltage, approimately
the constant output voltage is maintained with the presence o the large capacitive
flter. It is re/uently used or switching applications with a power in the range o
6:: to >:: .
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!i$erent types o topologies are there in which 1341 can be reali9ed such as Auc#
converter, Aoost converter, 1el ;scillating @y(bac# converter, Auc#(boost
converter, Aoost(buc#, %u#, 1epic. Aut only a ew are discussed in this article,
namely !% to !% converter, &% to !% converter, 0ly(bac# converter and 0orward
converter. 0or more inormation regarding the types o switch(mode power supply
and the types o 1341 with their wor#ing principles, eel ree to write your
comments or improving this article technically so that you can help the other
readers to get awareness o 1341.
Di!ital Data
In aCD (and any other digital recording technology), the goal is to create a recording with
veryhigh fidelity (very high similarity between the original signal and the reproduced
signal) andperfect reproduction (the recording sounds the same every single time you
play it no matter how many times you play it).
To accomplish these two goals, digital recording converts the analog wave into a stream of
numbers and records the numbers instead of the wave. The conversion is done by a device
called ananalog-to-digital converter (ADC). To play back the music, the stream of
numbers is converted back to an analog wave by adigital-to-analog converter (DAC). The
analog wave produced by the DAC is amplified and fed to thespeakers to produce the
sound.
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The analog wave produced by the DAC will be the same every time, as long as the
numbers are not corrupted. The analog wave produced by the DAC will also be
very similar to the original analog wave if the analog-to-digital converter sampled
at a high rate and produced accurate numbers.
You can understand why CDs have such high fidelity if you understand the analog-
to-digital conversion process better. Lets say you have a sound wave, and you wish
to sample it with an ADC. !ere is a typical wave "assume here that each tic# on the
hori$ontal a%is represents one-thousandth of a second&'
(hen you sample the wave with an analog-to-digital converter, you have control
over two variables'
• The samplin! rate ( %ontrols how many samples areta#en per second
• The samplin! precision ( %ontrols how many di$erent
gradations '/uanti9ation levels) are possible when ta#ing
the sample
)n the following figure, lets assume that the sampling rate is *,+++ per second and
the precision is *+'
The green rectangles represent samples. very one-thousandth of a second, the
ADC loo#s at the wave and pic#s the closest number between + and . The number
chosen is shown along the bottom of the figure. These numbers are a digital
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representation of the original wave. (hen the DAC recreates the wave from these
numbers, you get the blue line shown in the following figure'
You can see that the blue line lost uite a bit of the detail originally found in the
red line, and that means the fidelity of the reproduced wave is not very good. This
is the sampling error. You reduce sampling error by increasing both the sampling
rate and the precision. )n the following figure, both the rate and the precision have
been improved by a factor of / "/+ gradations at a rate of /,+++ samples per
second&'
)n the following figure, the rate and the precision have been doubled again "0+
gradations at 0,+++ samples per second&'
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You can see that as the rate and precision increase, the fidelity "the similarity
between the original wave and the DACs output& improves. )n the case of CD
sound, fidelity is an important goal, so the sampling rate is 00,*++ samples per
second and the number of gradations is 12,231. At this level, the output of the
DAC so closely matches the original waveform that the sound is essentially
4perfect4 to most human ears.
CD Stora!e Capacity
One thing about the CD's sampling rate and precision is that it produces a lot of data. On a
CD, the digital numbers produced by the ADC are stored asbytes, and it takes 2 bytes to
represent 65,536 gradations. There are two sound streams being recorded (one for each of
the speakers on a stereo system). A CD can store up to 74 minutes of music, so the total
amount of digital data that must be stored on a CD is:
44,100 samples/(channel*second) * 2 bytes/sample * 2 channels * 74 minutes * 60
seconds/minute = 783,216,000 bytes
With digital recording) audio engineers con"ert analog wa"es into digital
signals. There are many different kinds of e*uipment that can con"ert
analog to digital. Some audio studios record a performance on an analog
master tape first) then transfer the sound to a digital format. +thers will usespecial e*uipment to record directly to digital.
,arly digital recordings sacrificed fidelity) or sound *uality) in fa"or of
relia'ility. +ne of the draw'acks of an analog format is that analog media
tends to wear down. inyl al'ums can warp or get scratched) which can
dramatically impact sound *uality. Magnetic tape e"entually wears out and
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is "ulnera'le to magnets) which can erase or destroy information stored on
the tape. igital media like compact discs can reproduce sound indefinitely.
$nother ad"antage digital media has o"er analog is that you can make as
many copies of the original sound file as you like without hurting it.,"entually) e"en an analog master recording isnt going to sound as good
as the original performance. $s long as nothing corrupts a digital file) it will
stay the same no matter how much time has passed or how many copies
engineers make.
$nalog audio signals are suscepti'le to noise and distortion) due to the innate characteristics of electronic
circuits and associated de"ices. istur'ances in a digital system do not result in error unless the distur'ance is
so large as to result in a sym'ol 'eing misinterpreted as another sym'ol or distur' the se*uence of sym'ols. /t
is therefore generally possi'le to ha"e an entirely error-free digital audio system in which no noise or distortionis introduced 'etween con"ersion to digital format) and con"ersion 'ack to analog.
$ digital audio signal may 'e encoded for correction of any errors that might occur in the storage or
transmission of the signal) 'ut this is not strictly part of the digital audio process. This techni*ue) known
as 0channel coding0) is essential for 'roadcast or recorded digital systems to maintain 'it accuracy. The
discrete time and le"el of the 'inary signal allow a decoder to recreate the analog signal upon replay. ,ight to
&ourteen it Modulation is a channel code used in the audio !ompact isc 2!3.
Conversion process
The lifecycle of sound from its source) through an $!) digital processing) a $!) and finally as sound again.
$ digital audio system starts with an $! that con"erts an analog signal to a digital signal.4note 15 The $! runs at
a specified sampling rate and con"erts at a known 'it resolution. ! audio) for e6ample) has a sampling rate of
((.1 k782(()199 samples per second3) and has 1:-'it resolution for each stereo channel. $nalog signals thatha"e not already 'een'andlimited must 'e passed through an anti-aliasing filter 'efore con"ersion) to pre"ent
the distortion that is caused 'y audio signals with fre*uencies higher than the ;y*uist fre*uency) which is half
of the systems sampling rate.
$ digital audio signal may 'e stored or transmitted. igital audio can 'e stored on a !) a digital audio player )
a hard dri"e) a
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compression techni*ues) such as MP%) $d"anced $udio !oding) +gg or'is) or &=$!) are commonly
employed to reduce the file si8e. igital audio can 'e streamed to other de"ices.
&or play'ack) digital audio must 'e con"erted 'ack to an analog signal with a $!. $!s run at a specific
sampling rate and 'it resolution) 'ut may useo"ersampling) upsampling or downsampling to con"ert signals
that ha"e 'een encoded with a different sampling rate.
https://en.wikipedia.org/wiki/Audio_compression_(data)https://en.wikipedia.org/wiki/MP3https://en.wikipedia.org/wiki/Advanced_Audio_Codinghttps://en.wikipedia.org/wiki/Advanced_Audio_Codinghttps://en.wikipedia.org/wiki/Vorbishttps://en.wikipedia.org/wiki/Free_Lossless_Audio_Codechttps://en.wikipedia.org/wiki/Free_Lossless_Audio_Codechttps://en.wikipedia.org/wiki/Streaming_mediahttps://en.wikipedia.org/wiki/Oversamplinghttps://en.wikipedia.org/wiki/Upsamplinghttps://en.wikipedia.org/wiki/Upsamplinghttps://en.wikipedia.org/wiki/Downsamplinghttps://en.wikipedia.org/wiki/Downsamplinghttps://en.wikipedia.org/wiki/Audio_compression_(data)https://en.wikipedia.org/wiki/MP3https://en.wikipedia.org/wiki/Advanced_Audio_Codinghttps://en.wikipedia.org/wiki/Vorbishttps://en.wikipedia.org/wiki/Free_Lossless_Audio_Codechttps://en.wikipedia.org/wiki/Streaming_mediahttps://en.wikipedia.org/wiki/Oversamplinghttps://en.wikipedia.org/wiki/Upsamplinghttps://en.wikipedia.org/wiki/DownsamplingTop Related