Double Balanced Mixer

Double balanced mixers are able to provide very high levels of

performance in RF or frequency mixing applications.

The action of the double balanced mixer means that the input RF and
local oscillator signals are “balanced out” and their level is considerably
reduced at the output by having differential circuits on their inputs.

This reduces the need to remove the often unwanted RF and local
oscillator signals at the output and reduces the effect of these input
signals causing intermodulation distortion.

Double balanced mixers can either be made from the basic electronic
components, or they may also be bought as modules for inclusion in a
circuit - the latter approach is often adopted because the purchased
modules will have been designed, optimised and manufactured by
specialist manufacturers ensuing the highest performance.

In view of the level of their usage, double balanced mixers are widely
available from a number of specialist RF component suppliers. These
suppliers have a wide range of double balanced mixers both as hybrid
diode and FET based mixers as well as fully integrated MMIC based
devices that should meet the requirements for the majority of RF circuit
design applications.

Need for balanced mixers

Many forms of mixer are not balanced and as a result they allow through
considerable levels of the local oscillator and RF signals. These are normally
not wanted and normally they would have to be removed by filtering which is
often inconvenient and expensive. The solution is to balance the mixer to
remove the input signals.

There are two types of RF mixer that are balanced:

 ● Single balanced mixer: Often called just a balanced mixer, this
type of mixer will suppress either the LO or RF signal but not both
● Double balanced mixer Unlike the single balanced mixer, the
double balanced mixer suppresses both of the input signals.

While single balanced mixers offer many advantages over simpler designs,
the double balanced mixer is more widely used. However there are a number
of advantages and disadvantages over a single balanced mixer to consider:

Double balanced mixer advantages.

● Increased linearity.
● Better suppression of spurious products - all even order products of
the LO and RF inputs are suppressed.
● Isolation between all ports.

RF / frequency mixer ports

Like all other RF mixers, double balanced mixers have the same three ports or
connections.

● RF input: This port on the mixer is connected to the incoming signal that
is to have its frequency converted.
● Local Oscillator or LO input: This port takes in the internal local
oscillator signal that is used to convert the RF signal to the new frequency.
● IF output: The third port of the double balanced mixer is normally referred
to as the IF or intermediate frequency output. The signal on the output of
an ideal RF mixer should contain only the mixer products, i.e. the sum and
difference frequencies of the two input signals.
●
Types of double balanced mixer
Double balanced mixers come in a variety of forms using different types of
electronic component and having some slightly different formats.

● Hybrid based diode double balanced mixer: This type of double

balanced mixer is the stereotypes double balanced mixer. It uses two
main electronic components namely a ring of four diodes, which are
normally Schottky diodes and baluns on two or more ports. It is
primarily these baluns that add the cost and also limit the frequency
response of the RF mixer.
● Hybrid based active double balanced mixer: This form of double
balanced mixer replaces the diodes with an active device to act as
the switching elements within the ring circuit. Again hybrid baluns are
used on both input circuits, but the format is again the same. The
electronic components include the expensive wound baluns as well
as the active devices, e.g. FETs which have a good switching
performance.
● Active double balanced mixer: Using active differential amplifier
circuits, it is possible to achieve the balanced operation for the RF
mixer. This enables the complete mixer circuit to be fabricated on a
single semiconductor chip. Most high performance RF mixers, as well
as many lower performance ones use this technology. The cost of the
high performance active double balanced mixers is much less than
 those using wound components, and they offer a much wider
bandwidth.

In view of the better RF performance and lower cost of active double balanced
mixers using no hybrids, this type of mixer is used in most RF circuit design
applications these days. Mixers are available in standard surface mount
technology packages for many applications and for more exacting ones,
special monolithic microwave ICs, MMICs are available for RF mixers.

Reversing switch mixers

Double balanced mixers are a form of what is termed a "reversing switch mixer."
Reversing switch mixers operate by using electronic switches in a bridge formation
to reverse the input RF signal under the action of the local oscillator used as a
square wave switching signal. They normally offer significant advantages over
analogue mixers for radio communications and general RF design applications as
they are able to offer better levels of dynamic range and noise. In view of this fact,
they are normally used in high performance applications where noise and dynamic
range are of importance - e.g. in the front end of a radio receiver or spectrum
analyzer.

Double balanced mixer basics

The most common form of double balanced mixer is the diode double balanced
mixer. In its simplest form it consists of two unbalanced to balanced transformers
and a diode ring consisting of four diodes as shown.
Although the design of the RF mixer looks straightforward, high performance mixers
are designed and built to exacting standards to achieve the high levels of
performance needed.

One of the key specifications for a double balanced mixer is whether any of the LO
or RF signals appear at the IF port. This depends upon the diode and transformer
uniformity. In addition to this the circuit offers high isolation between the RF and IF
ports because the balanced diode switching precludes direct connection between T1
and T2.

Double balanced mixer

components
Although there are comparatively few components in a double balanced
mixer, their individual performance is crucial to the performance of the RF
mixer as a whole.

Normally Schottky barrier diodes are used for the diode ring. They offer a low
on resistance and they also have a good high frequency response. Ordinary
signal diodes may be used for low performance applications, although the
cost difference is small. It is found that the diode forward voltage drop for the
diodes determines the optimum local oscillator drive level. RF mixers requiring
to handle a high RF input level will need a correspondingly high LO input
level. As a rule of thumb the LO signal level should be a minimum of 20dB
higher than either the RF or IF signals. This ensures that the LO signal rather
than the RF or IF signals switch the RF mixer, and this is a key element in
reducing intermodulation distortion, IMD, and also maximising the dynamic
range.

To increase the required drive level, it is possible to place multiple diodes in

each leg. The most common LO drive level for a double balanced mixer is
probably +7dBm. However they can be obtained with a variety of drive levels.
Values of 0, +3, +7, +10, +13, +17, +23, and +27 dBm are normally available.

In order to provide the required level of performance, the quad diodes used
win these mixers are generally fabricated monolithically. By doing this they will
have very closely matched performance parameters, and in particular the
level of forward voltage will be virtually identical in all the diodes.

The transformers are also critical to the performance of the RF mixer. Creating
a wideband balun for the mixer is one of the key elements within the overall
mixer design and achieving the required bandwidth and performance can be
difficult to achieve.

The matching of the transformers and the individual legs are important in
determining the balance of the RF mixer. The transformer also plays an
important role in determining the conversion loss and drive level of the RF
mixer. As the transformers are wound on a ferrite core, the core loss, copper
loss and impedance mismatch all contribute to the transformer losses.

Double balanced mixer

operation
The operation of the double balanced mixer is relatively easy to understand.
The local oscillator, LO, signal turns on first one arm (D3, D4), and then the
other (D1, D2) within the diode ring.
As the points where the LO signal enters the diode ring at the junction of D1
and D4 appear as a virtual earth to the RF signal, this means that the points
where the RF signal enters are alternatively connected to ground as the
diodes turn on anThe operation of the mixer means that the RF signal with
alternating inverse phases is routed to the IF port according to the switching
action of the local oscillator - in other words the signal at the IF port has been
multiplied by the local oscillator waveform.

Double balanced FET mixer

While diode mixers are able to offer excellent performance, the increase in
use of wireless and general radio communications systems means that
receivers need to be able to accommodate a larger number of local strong
signals than may have been the case previously. Better low end noise
performance along with higher third order intercept points are required. The
performance figures required by double balanced diode mixers cannot always
meet the requirements for some designs, unless significant tailoring is
undertaken and this increases the costs beyond economic viability.
Conventional double balanced diode mixers can offer a third order intercept
performance up to figures of between about +25 and +30 dBm.

To offer an alternative to the diode mixer, it is possible to use a double

balanced FET mixer. Well-designed FET mixers are able to offer extremely
linear performance along with high third order intercept points - some as high
as +38dBm.
The diagram shows the basic concept of a double balanced FET mixer.
However some mixers require the application of a DC bias to ensure the
correct switching of the diodes, and some mixers show a high
conversion loss or noise figure. Double balanced FET mixers using
discrete components can sometimes be optimised to provide better
performance figures, and newer commercially available items are also
offering better performance.

Linearization Techniques

Power amplifiers are indispensable components in a communication

system and are inherently nonlinear. To reduce the nonlinearity, the power
amplifier can be backed off to operate within the linear portion of its
operating curve. To improve the power amplifier efficiency without
compromising its linearity, power amplifier linearization is essential. Various
linearization techniques are used for linearity and efficiency improvement in
mobile communication systems. Among all linearization techniques, digital
predistortion is one of most important technique. Power amplifier’s output
suffers from noise especially when used for prolonged hours. The heat
dissipated during the operation normally violates the faithful amplification of
the input signal at desired gain. The heat dissipation is inherent in the
system due to long run and can not be avoided, however can be
compensated by using appropriate algorithm in order to keep track of
faithful amplification.

Power amplifier linearization is currently one of the most promising

techniques for linearity and efficiency improvement in mobile
communication systems. There are numerous techniques which have
different levels of complexity, various advantages and limitations. Different
linearization methods may fit to different communication systems.

POWER AMPLIFIERS FEATURES

Gain and Output Power:

In mobile communications each system has its specifications which must

be fulfilled. Obtaining output powers high enough for various applications is
a very important task achieved by Power Amplifiers. In general the
information signal is first modulated and up converted, and then sent to a
PA. This input is multiplied with a gain factor and the desired output power
is obtained.

Linearity:

Linearity is one of the key issues in Power Amplifiers used in new

generation mobile communication systems. The linearity of a Power
Amplifier is easily visible in its gain and phase characteristics.There are
many linearization methods used for power amplifiers which give better
linearity but at the cost of poor efficiency i.e. the method which give good
linearity with efficiency should be used for linearization of power
amplifiers.If an amplifier has a constant gain and phase response for an
input power region, then the amplifier is said to be linear for this region.

Efficiency:

Efficiency is another key issue in mobile communications especially for

battery operated mobile terminals. It has two widely used definitions, drain
(or collector) efficiency and PAE (Power Added Efficiency). Drain efficiency
is the ratio of output radio frequency (RF) power to input DC power

η = Pout RF /PDC

LINEARIZATION METHOD

FEEDBACK METHOD

Feedback linearization methods are relatively simple compared to

feedforward and conventional predistortion. The idea is to force the PA
output to follow its input. There are different types of feedback linearization
topologies classified mainly as RF feedback and modulation feedback
which can be divided again into two: polar and Cartesian feedback. In
modulation feedback the modulation components (I&Q or R&θ) of PA input
and output are compared whereas in RF feedback the RF signals are
compared. Feedback systems can be implemented at RF, IF or baseband
frequencies. A major issue in feedback linearization is the stability due to
delays in feedback which is critical, especially in systems with discrete
components. The group delay increases significantly due to PA matching
circuits or filters and couplers in the loop.

FEEDFORWARD METHOD

The feed forward method allows high linearization performance and is

currently used in base stations of mobile communications systems.

The idea is to extract the distortion at the PA output, amplify it and add it to
the PA output in opposite phase in order to cancel the distortion. Out of all
linearization methods, only feedforward systems provide a very good
distortion reduction over a wide bandwidth. The drawback of these systems
is the low power efficiency due to high power requirement of the error
amplifier operated in class A mode and losses due to couplers and delay
lines in the system.

PREDISTORTION METHOD
The idea behind predistortion is to expand the input signal prior a PA in
such a way that the nonlinearities due to the PA are compensated. It is
realized by implementing a nonlinear block in front of the nonlinear PA
generating input signal level dependent distortion elements opposite of the
distortion caused by the PA. As a result the cascade of these nonlinear
blocks has a linear response.

Power amplifier linearization techniques are very important to reduce the

distortion of the transmitted signal and the adjacent band interference of
users. As its facility of implementation, adaptive ability and high efficiency,
the pre-distortion technology becomes the first choice to minimize the
nonlinear distortions.
Amplifier Classes
On the basis of the mode of operation, i.e., the portion of the input

cycle during which collector current flows, the power amplifiers may

be classified as follows. Power amplifier circuits (output stages) are

classified as A, B, AB and C for linear designs—and class D and E for

switching designs.

● Class A Power amplifier − When the collector current flows at

all times during the full cycle of signal, the power amplifier is

known as class A power amplifier.

● Class B Power amplifier − When the collector current flows

only during the positive half cycle of the input signal, the power

amplifier is known as class B power amplifier.

● Class C Power amplifier − When the collector current flows for

less than half cycle of the input signal, the power amplifier is

known as class C power amplifier.

There forms another amplifier called Class AB amplifier, if we

combine the class A and class B amplifiers so as to utilize the

advantages of both.
Terms Considering Performance

The primary objective of a power amplifier is to obtain maximum output power. In

order to achieve this, the important factors to be considered are collector efficiency,

power dissipation capability and distortion. Let us go through them in detail.

Collector Efficiency
This explains how well an amplifier converts DC power to AC power. When the DC

supply is given by the battery but no AC signal input is given, the collector output at

such a condition is observed as collector efficiency.

The collector efficiency is defined as

η= average a.c power output / average d.c power input to transistor

For example, if the battery supplies 15W and AC output power is 3W. Then the

transistor efficiency will be 20%.

The main aim of a power amplifier is to obtain maximum collector efficiency. Hence the

higher the value of collector efficiency, the efficient the amplifier will be.

Power Dissipation Capacity

Every transistor gets heated up during its operation. As a power transistor handles

large currents, it gets more heated up. This heat increases the temperature of the

transistor, which alters the operating point of the transistor.

So, in order to maintain the operating point stability, the temperature of the transistor

has to be kept in permissible limits. For this, the heat produced has to be dissipated.

Such a capacity is called as Power dissipation capability.

Power dissipation capability can be defined as the ability of a power transistor to

dissipate the heat developed in it. Metal cases called heat sinks are used in order to

dissipate the heat produced in power transistors.

Distortion
A transistor is a non-linear device. When compared with the input, there occur few

variations in the output. In voltage amplifiers, this problem is not pre-dominant as

small currents are used. But in power amplifiers, as large currents are in use, the

problem of distortion certainly arises.

Distortion is defined as the change of output wave shape from the input wave shape of

the amplifier. An amplifier that has lesser distortion, produces a better output and

hence considered efficient.

Class A Power Amplifiers
Class A amplifier is a high gain amplifier with high linearity. In case of Class A
amplifier, the conduction angle is 360 degree. As we stated above, a 360-degree
conduction angle means the amplifier device remains active for the entire
time and use complete input signal. In the below image an ideal class A
amplifier is shown. As we can see in the image, there is one active element, a
transistor. The bias of the transistor remains ON all of the time. Due to this never
turn off feature, Class A amplifier provides better high frequency and
feedback loop stability. Other than these advantages, Class A amplifier is easy
to construct with a single-device component and minimum parts count.
Despite the advantages and high linearity, certainly, it has many limitations. Due
to continuous conducting nature, the class A amplifier introduce high power
loss. Also, due to high linearity, Class A amplifier provides distortion and noises.
The power supply and the bias construction need careful component selection to
avoid unwanted noise and to minimize the distortion.
Because of high power loss in Class A amplifier, it emits heat and requires higher
heat sink space. The efficiency is very poor in Class A amplifiers, theoretically,
the efficiency varies between 25 to 30 % if used with the usual configuration. The
efficiency can be improved using inductively coupled configuration but the
efficiency in such case is not more than 45-50%, thus it is only suitable for low
signal or low power level amplification purposes.

The output characteristics with operating point Q is shown in the figure above. Here

(Ic)Q and (Vce)Q represent no signal collector current and voltage between collector and

emitter respectively. When signal is applied, the Q-point shifts to Q1 and Q2. The

output current increases to (Ic)max and decreases to (Ic)min. Similarly, the

collector-emitter voltage increases to (Vce)max and decreases to (Vce)min.

D.C. Power drawn from collector battery Vcc is given by

Pin =voltage×current= VCC(IC)Q

(η)overall = a.c power delivered to the load / total power delivered by d.c
supply

(η)collector=average a.c power output / average d.c power input to transistor

Advantages of Class A Amplifiers

The advantages of Class A power amplifier are as follows −

​ The current flows for complete input cycle

​ It can amplify small signals
​ The output is same as input
​ No distortion is present

Disadvantages of Class A Amplifiers

The disadvantages of Class A power amplifier are as follows −

​ Low power output

​ Low collector efficiency

Class B Amplifier
The Class B amplifier is a bit different from the Class A. It is created using
two active devices which conduct half of the actual cycle, ie 180 degrees
of the cycle. Two devices provide combined current drive for the load.
In the above image, an Ideal Class B amplifier configuration has been shown.
It consists two active devices which get biased one by one during the
positive and negative half cycle of sinusoidal wave and thus the signal gets
pushed or pulled to the amplified level from both positive and negative side
and combine the result we get complete cycle across the output. Each device
turned on or became active half of the cycle, and due to this the efficiency
gets improved, comparing to 25- 30% efficiency of Class A amplifier, it
provides more than 60% efficiency theoretically. We can see each device
input and output signal graph in the below image. The efficiency is not more
than 78% for Class B amplifier. The heat dissipation is minimized in this
class providing a low heat sink space.
But, this class also have limitation. A very profound limitation of this class
is the crossover distortion. As two devices provides each half of the
sinusoidal waves which are combined and joined across the output, there is a
mismatch (cross over) in the region, where two halves are combined. This is
because when one device complete the half cycle, the other one needs to
provide the same power almost at the same time when other one finish the
job. It is difficult to fix this error in class A amplifier as during the active
device the other device remains completely inactive. The error provides a
distortion in the output signal. Due to this limitation, it is a major fail for
precision audio amplifier application.
Advantages
The advantages of Complementary symmetry push pull class B

amplifier are as follows.

● As there is no need of center tapped transformers, the

weight and cost are reduced.

● Equal and opposite input signal voltages are not required.

Disadvantages
The disadvantages of Complementary symmetry push pull class

B amplifier are as follows.

● It is difficult to get a pair of transistors (NPN and PNP) that

have similar characteristics.

● We require both positive and negative supply voltages.

Class AB Amplifier
An alternate approach to overcome the cross-over distortion, is to use the AB

amplifier. Class AB amplifier uses intermediate conduction angle of both

Classes A and B, thus we can see the property of both Class A and Class B

amplifier in this AB class of amplifier topology. Same as class B, it has the same

configuration with two active devices which conducts during half of the cycles
individually but each device biased differently so they do not get completely

OFF during the unusable moment (crossover moment). Each device does not

leave the conduction immediately after completing the half of the sinusoidal

waveform, instead they conduct a small amount of input on another half cycle.

Using this biasing technique, the crossover mismatch during the dead zone is

dramatically reduced.

But in this configuration, efficiency is reduced as the linearity of the devices is

compromised. The efficiency remains more than the efficiency of typical Class A

amplifier but it is less than the Class B amplifier system.

Also, the diodes need to be carefully chosen with the exact same rating and

need to be placed as close as possible to the output device. In some circuit

construction, designers tend to add small value resistor to provide stable

quiescent current across the device to minimize the distortion across the output.

Class C Amplifier
Apart from the Class A, B, and AB amplifier, there is another amplifier Class C.

It’s a traditional amplifier which works differently than the other amplifiers

classes. Class C amplifier is tuned amplifier which works in two different

operating modes, tuned or untuned. The efficiency of Class C amplifier is much

more than the A, B, and AB. Maximum 80% efficiency can be achieved in radio

frequency related operations

Class C amplifier uses less than 180-degree conduction angle. During the

untuned mode, the tuner section is omitted from the amplifier configuration. In

this operation, Class C amplifier also gives huge distortion across the output.

When the circuit is exposed to a tuned load, the circuit clamps the output bias

level with the average output voltage equal to the supply voltage. The tuned

operation is called as clamper. During this operation, the signal gets its proper

shape and the center frequency became less distorted.

In typical uses, Class C amplifier gives 60-70% efficiency.

Direct Digital Synthesis DDS
Direct digital synthesis, DDS, is a frequency synthesizer technique that is
becoming more widespread.

DDS, direct digital synthesis takes a different approach to that of the more
usual indirect frequency synthesis techniques using PLLs by directly
synthesising the waveform from a digital map of the waveform stored in a
memory.

Using digital techniques in this way, along with high speed logic, direct digital
synthesis provides a powerful technique for creating accurate signals whose
frequency can be stepped by very small increments giving virtually analogue
or continuous tuning if needed.
For many years, direct digital synthesizers were limited in frequency by the
speed of the logic. With speeds improving he top frequency limits for direct
digital synthesizers is increasing.

How DDS works

As the name suggests this form of synthesis generates the waveform directly
using digital techniques. This is different to the way in which the more familiar
indirect synthesizers that use a phase locked loop as the basis of their
operation.

A direct digital synthesizer operates by storing the points of a waveform in

digital format, and then recalling them to generate the waveform. The rate at
which the synthesizer completes one waveform then governs the frequency.
The overall block diagram is shown below, but before looking at the details
operation of the synthesizer it is necessary to look at the basic concept behind
the system.

The operation can be envisaged more easily by looking at the way that phase
progresses over the course of one cycle of the waveform. This can be
envisaged as the phase progressing around a circle. As the phase advances
around the circle, this corresponds to advances in the waveform
The synthesizer operates by storing various points in the waveform in digital
form and then recalling them to generate the waveform. Its operation can be
explained in more detail by considering the phase advances around a circle
as shown in Figure 2. As the phase advances around the circle this
corresponds to advances in the waveform, i.e. the greater the number
corresponding to the phase, the greater the point is along the waveform. By
successively advancing the number corresponding to the phase it is possible
to move further along the waveform cycle.

The digital number representing the phase is held in the phase accumulator.
The number held here corresponds to the phase and is increased at regular
intervals. In this way it can be sent hat the phase accumulator is basically a
form of counter. When it is clocked it adds a preset number to the one already
held. When it fills up, it resets and starts counting from zero again. In other
words this corresponds to reaching one complete circle on the phase diagram
and restarting again.

Once the phase has been determined it is necessary to convert this into a
digital representation of the waveform. This is accomplished using a waveform
map. This is a memory which stores a number corresponding to the voltage
required for each value of phase on the waveform. In the case of a
synthesizer of this nature it is a sine look up table as a sine wave is required.
In most cases the memory is either a read only memory (ROM) or
programmable read only memory (PROM). This contains a vast number of
points on the waveform, very many more than are accessed each cycle. A
very large number of points is required so that the phase accumulator can
increment by a certain number of points to set the required frequency.

The next stage in the process is to convert the digital numbers coming from
the sine look up table into an analogue voltage. This is achieved using a
digital to analogue converter (DAC). This signal is filtered to remove any
unwanted signals and amplified to give the required level as necessary.

Tuning is accomplished by increasing or decreasing the size of the step or

phase increment between different sample points. A larger increment at each
update to the phase accumulator will mean that the phase reaches the full
cycle value faster and the frequency is correspondingly high. Smaller
increments to the phase accumulator value means that it takes longer to
increase the full cycle value and a correspondingly low value of frequency. In
this way it is possible to control the frequency. It can also be seen that
frequency changes can be made instantly by simply changing the increment
value. There is no need to a settling time as in the case of phase locked loop
based synthesizer.

From this it can be seen that there is a finite difference between one
frequency and the next, and that the minimum frequency difference or
frequency resolution is determined by the total number of points available in
the phase accumulator. A 24 bit phase accumulator provides just over 16
million points and gives a frequency resolution of about 0.25 Hz when used
with a 5 MHz clock. This is more than adequate for most purposes.
These synthesizers do have some disadvantages. There are a number of
spurious signals which are generated by a direct digital synthesizer. The most
important of these is one called an alias signal. Here images of the signal are
generated on either side of the clock frequency and its multiples. For example
if the required signal had a frequency of 3 MHz and the clock was at 10 MHz
then alias signals would appear at 7 MHz and 13 MHz as well as 17 MHz and
23 MHz etc.. These can be removed by the use of a low pass filter. Also some
low level spurious signals are produced close in to the required signal. These
are normally acceptable in level, although for some applications they can
cause problems.

What are the main benefits of using a DDS?

DDS devices like the AD9833 are programmed through a high speed serial
peripheral-interface (SPI), and need only an external clock to generate
simple sine waves. DDS devices are now available that can generate
frequencies from less than 1 Hz up to 400 MHz (based on a 1-GHz clock).
The benefits of their low power, low cost, and single small package,
combined with their inherent excellent performance and the ability to
digitally program (and reprogram) the output waveform, make DDS devices
an extremely attractive solution—preferable to less-flexible solutions
comprising aggregations of discrete elements.