The superhet or superheterodyne radio receiver

- an introduction to the operation of the superheterodyne radio receiver and how it uses the process of mixing and frequency translation with an intermediate frequency amplifier and filter to provide high levels of selectivity and amplification

The superhet radio or to give it its full name the superheterodyne receiver is one of the most popular forms of receiver in use today. Virtually all broadcast radios, televisions and many more types of receiver use the superhet or superheterodyne principle. First developed at the end of the First World War, with its invention credited to the American Edwin Armstrong, the use of the superhet has grown ever since the concept was first discovered.

MixingThe idea of the superhet revolves around the process of mixing. Here RF mixers are used to multiply two signals together. (This is not the same as mixers used in audio desks where the signals are added together). When two signals are multiplied together the output is the product of the instantaneous level of the signal at one input and the instantaneous level of the signal at the other input. It is found that the output contains signals at frequencies other than the two input frequencies. New signals are seen at frequencies that are the sum and difference of the two input signals, i.e. if the two input frequencies are f1 and f2, then new signals are seen at frequencies of (f1+f2) and (f1-f2). To take an example, if two signals, one at a frequency of 5 MHz and another at a frequency of 6 MHz are mixed together then new signals at frequencies of 11 MHz and 1 MHz are generated.

The signals generated by mixing or multiplying two signals together

Concept of the superheterodyne receiverIn the superhet or superheterodyne radio, the received signal enters one input of the mixed. A locally generated signal (local oscillator signal) is fed into the other. The result is that new signals are generated. These are applied to a fixed frequency intermediate frequency (IF) amplifier and filter. Any signals that are converted down and then fall within the passband of the IF amplifier will be amplified and passed on to the next stages. Those that fall outside the passband of the IF are rejected. Tuning is accomplished very simply by varying the frequency of the local oscillator. The advantage of this process is that very selective fixed frequency filters can be used and these far out perform any variable frequency ones. They are also normally at a lower frequency than the incoming signal and again this enables their performance to be better and less costly.

To see how this operates in reality take the example of two signals, one at 6 MHz and another at 6.1 MHz. Also take the example of an IF situated at 1 MHz. If the local oscillator is set to 5 MHz, then the two signals generated by the mixer as a result of the 6 MHz signal fall at 1 MHz and 11 MHz. Naturally the 11 MHz signal is rejected, but the one at 1 MHz passes through the IF stages. The signal at 6.1 MHz produces a signal at 1.1 MHz (and 11.1 MHz) and this falls outside bandwidth of the IF so the only signal to pass through the IF is that from the signal on 6 MHz.

The basic concept of the superhet radio

If the local oscillator frequency is moved up by 0.1 MHz to 5.1 MHz then the signal at 6.1 MHz will give rise to a signal at 1 MHz and this will pass through the IF. The signal at 6 MHz will give rise to a signal of 0.9 MHz at the IF and will be rejected. In this way the receiver acts as a variable frequency filter, and tuning is accomplished.

ImagesThe basic concept of the superheterodyne receiver appears to be fine, but there is a problem. There are two signals that can enter the IF. With the local oscillator set to 5 MHz and with an IF it has already been seen that a signal at 6 MHz mixes with the local oscillator to produce a signal at 1 MHz that will pass through the IF filter. However if a signal at 4 MHz enters the mixer it produces two mix products, namely one at the sum frequency which is 10 MHz, whilst the difference frequency appears at 1 MHz. This would prove to be a problem because it is perfectly possible for two signals on completely different frequencies to enter the IF. The unwanted frequency is known as the image. Fortunately it is possible to place a tuned circuit before the mixer to prevent the signal entering the mixer, or more correctly reduce its level to acceptable value.

Fortunately this tuned circuit does not need to be very sharp. It does not need to reject signals on adjacent channels, but instead it needs to reject signals on the image frequency. These will be separated from the wanted channel by a frequency equal to twice the IF. In other words with an IG at 1 MHz, the image will be 2 MHz away from the wanted frequency.

Using a tuned circuit to remove the image signal

Complete receiverHaving looked at the concepts behind the superheterodyne receiver it is helpful to look at a block diagram of a basic superhet. Signals enter the front end circuitry from the antenna. This contains the front end tuning for the superhet to remove the image signal and often includes an RF amplifier to amplify the signals before they enter the mixer. The level of this amplification is carefully calculated so that it does not overload the mixer

when strong signals are present, but enables the signals to be amplified sufficiently to ensure a good signal to noise ratio is achieved.

The tuned and amplified signal then enters one port of the mixer. The local oscillator signal enters the other port. The local oscillator may consist of a variable frequency oscillator that can be tuned by altering the setting on a variable capacitor. Alternatively it may be a frequency synthesizer that will enable greater levels of stability and setting accuracy.

Once the signals leave the mixer they enter the IF stages. These stages contain most of the amplification in the receiver as well as the filtering that enables signals on one frequency to be separated from those on the next. Filters may consist simply of LC tuned transformers providing inter-stage coupling, or they may be much higher performance ceramic or even crystal filters, dependent upon what is required.

Once the signals have passed through the IF stages of the superheterodyne receiver, they need to be demodulated. Different demodulators are required for different types of transmission, and as a result some receivers may have a variety of demodulators that can be switched in to accommodate the different types of transmission that are to be encountered. The output from the demodulator is the recovered audio. This is passed into the audio stages where they are amplified and presented to the headphones or loudspeaker.

Block diagram of a basic superheterodyne receiver

The diagram above shows a very basic version of the superhet or superheterodyne receiver. Many sets these days are far more complicated. Some superhet radios have more than one frequency conversion, and other areas of additional circuitry to provide the required levels of performance. However the basic superheterodyne concept remains the same, using the idea of mixing the incoming signal with a locally generated oscillation to convert the signals to a new frequency.

 Selectivity is one of the major specifications of any receiver. Whilst the sensitivity is important to ensure that it can pick up the signals and receive them at a sufficient strength, the selectivity is also very important. It is this parameter that determines whether the receiver is able to pick out the wanted signal from all the other ones around it. The filters used in receivers these days have very high levels of performance and enable receivers to select out individual signals even on today's crowded bands.

Superhet principleMost of the receivers that are used today are superhet radios. In these sets the incoming signal is converted down to a fixed intermediate frequency. It is within the IF stages that the main filters are to be found. It is the filter in the IF stages that defines the selectivity performance of the whole set, and as a result the receiver selectivity specification is virtually that of the filter itself.

Block diagram of a basic superhet receiver

In some receivers simple LC filters may be used, although ceramic filters are better and are used more widely nowadays. For the highest performance crystal or mechanical filters may be used, although they are naturally more costly and this means they are only found in high performance sets.

Filter parametersThere are two main areas of interest for a filter, the pass band where it accepts signals and allows them through, and the stop band where it rejects them. In an ideal world a filter would have a response something like that shown in Figure 2. Here it can be seen that there is an immediate transition between the pass band and the stop band. Also in the pass band the filter does not introduce any loss and in the stop band no signal is allowed through.

The response of an ideal filter

In reality it is not possible to realise a filter with these characteristics and a typical response more like that shown in Figure 3. It is fairly obvious from the diagram that there are a number of differences. The first is that there is some loss in the pass band. Secondly the response does not fall away infinitely fast. Thirdly the stop band attenuation is not infinite, even though it is very large. Finally it will be noticed that there is some in band ripple.

Typical response of a real filter

In most filters the attenuation in the pass band is normally relatively small. For a typical crystal filter figures of 2 - 3 dB are fairly typical. However it is found that very narrow band filters like those used for Morse reception may be higher than this. Fortunately it is quite easy to counteract this loss simply by adding a little extra amplification in the intermediate frequency stages and this factor is not quoted as part of the receiver specification.

It can be seen that the filter response does not fall away infinitely fast, and it is necessary to define the points between which the pass band lies. For receivers the pass band is taken to be the bandwidth between the points where the response has fallen by 6 dB, i.e. where it is 6 dB down or -6 dB.

A stop band is also defined. For most receiver filters this is taken to start at the point where the response has fallen by 60 dB, although the specification for the filter should be checked this as some filters may not be as good. Sometimes a filter may have the stop band defined for a 50 dB attenuation rather than 60 dB.

Shape factorIt can be seen that it is very important for the filter to achieve its final level of rejection as quickly as possible once outside the pass band. In other words the response should fall as quickly as possible. To put a measure on this, a figure known as the shape factor is used. This is simply a ratio of the bandwidths of the pass band and the stop band. Thus a filter with a pass band of 3 kHz at -6dB and a figure of 6 kHz at -60 dB for the stop band would have a shape factor of 2:1. For this figure to have real meaning the two attenuation figures should also be quoted. As a result the full shape factor specification should be 2:1 at 6/60 dB.

Filter typesThere is a variety of different types of filter that can be used in a receiver. The older broadcast sets used LC filters. The IF transformers in the receiver were tuned and it was possible to adjust the resonant frequency of each transformer using an adjustable ferrite core.

Today ceramic filters are more widely used. Their operation is based on the piezoelectric effect. The incoming electrical signal is converted into mechanical vibrations by the piezoelectric effect. These vibrations are then affected by the mechanical resonances of the ceramic crystal. As the mechanical vibrations are then linked back to the electric signal, the overall effect is that the mechanical resonances of the ceramic crystal affect the electrical signal. The mechanical resonances of the ceramic exhibit a high level of Q and this is reflected in its performance as an electrical filter. In this way a high Q filter can be manufactured very easily.

Ceramic filters can be very cheap, some costing only a few cents. However higher performance ones are also available, and these are likely to be found in scanners and many other receivers.

For really high levels of filter performance crystal filters are used. Crystals are made from quartz, a naturally occurring form of silicon, although today's components are made from synthetically grown quartz. These crystals also use the piezoelectric effect and operate in the same way as ceramic filters but they exhibit much higher levels of Q and offer far superior degrees of selectivity. Being a resonant element they are used in many areas where an LC resonant element might be found. They are used in oscillators - many computers have crystal oscillators in them, but they are also widely used in high performance filters.

Normally crystal filters are made from a number of individual crystals. The most commonly used configuration is called the half lattice filter as shown in Figure 4. Further sections can be added to the filter to improve the performance. Often a filter will be quoted as having a certain number of poles. There is one pole per crystal, so a six pole crystal filter would contain six crystals and so forth. Many filters used in amateur communications receivers will contain either six or eight poles.

A basic half lattice crystal filter section

Choosing the right bandwidthIt is important to choose the correct bandwidth for a give type of signal. It is obviously necessary to ensure that it is not too wide, otherwise unwanted off-channel signals will be able to pass though the filter. Conversely if the filter is too narrow then some of the wanted signal will be rejected and distortion will occur. As different types of transmission occupy different amounts of spectrum bandwidth it is necessary to tailor the filter bandwidth to the type of transmission being received. As a result many receivers switch in different filters for different types of transmission. This may be done either automatically as part of a mode switch, or using a separate filter switch. Typically a filter for AM reception on the short wave bands will have a bandwidth of around 6 kHz, and one for SSB will be approximately 2.5 kHz. For Morse reception 500 and 250 Hz filters are often used.

SummarySelectivity is particularly important on today's crowded bands, and it is necessary to ensure that any receiver is able to select the wanted signal as well as it can. Obviously when signals occupy the same frequency there is little that can be done, but by having a good filter it is possible to ensure that you have the best chance or receiving and being able to copy the signal you want.

 he superhet radio receiver is one of the most widely used types of receiver available. One of the important specifications associated with its operation is image response or image rejection. Along with this the IF breakthrough is also of importance, although less critical in many applications.

Image responseThe basic concept of the superhet radio means that it is possible for two signals to eneter the intermediate frequency (IF) implifier. For example with the local oscillator set to 5 MHz and with an IF of 1 MHz it can be seen that a signal at 6 MHz mixes with the local oscillator to produce a signal at 1 MHz that will pass through the IF filter. However is a signal at 4 MHz is also able to produce an output at 1 MHz. It is clearly unacceptable to receive signals on two frequencies at the same time and it is possible to remove the unwanted one by the addition of a tuned circuit prior to the mixer

Fortunately this tuned circuit does not need to be excessively sharp. It does not need to reject signals on adjacent channels, but instead it needs to reject signals on the image frequency. These will be separated from the wanted channel by a frequency equal to twice the IF. In other words with an IG at 1 MHz, the image will be 2 MHz away from the wanted frequency.

Using a tuned circuit to remove the image signal

ImageIt is clearly important to specify the level of rejection of the image signal. The specification compares the levels of signals of equal strength on the wanted and image frequencies, quoting the level of rejection of the unwanted signal.

The image rejection of a receiver will be specified as the ratio between the wanted and image signals expressed in decibels (dB)at a certain operating frequency. For example it may be 60 dB at 30 MHz. This means that if signals of the same strength were present on the wanted frequency and the image frequency, then the image signal would be 60 dB lower than the wanted one, i.e. it would be 1/1000 lower in terms of voltage or 1/1000000 lower in terms of power.

The frequency at which the measurement is made also has to be included. This is because the level of rejection will vary according to the frequency in use. Typically it falls with increasing frequency because the percentage frequency difference between the wanted and image signals is smaller.

IF BreakthroughAnother problem which can occur with a superhet occurs when signals from the aerial break through the RF sections of the set and directly enter the IF stages. Normally intermediate frequencies are chosen so that there are likely to be no very large signals present which might cause problems. However when the receiver has a fixed frequency first local oscillator this is not easy to ensure as it will sweep over a band of frequencies.

The specification for breakthrough is quoted in the same fashion as image rejection. Normally it is possible to achieve figures of 60 to 80 dB rejection, and on some receivers figures of 100 dB have been quoted.

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Radio receiver sensitivity

- including the concept of noise and sensitivity, signal to noise ratio, SINAD, and noise figure.

Receiver sensitivity is one of the key specifications of any radio. The two main requirements of any radio receiver are that it should be able to separate one station from another, i.e. selectivity, and signals should be amplified so that they can be brought to a sufficient level to be heard. As a result receiver designers battle with many elements to make sure that these requirements are fulfilled

A number of methods of measuring and specifying the sensitivity performance of radio receivers are used. Figures including signal to noise ratio, SINAD, noise factor and noise figure are used. These all use the fact that the limiting factor of the sensitivity of a radio receiver is not the level of amplification available, but the levels of noise that are present, whether they are generated within the radio receiver or outside it.

NoiseToday technology is such that there is little problem in being able to achieve very large levels of amplification within a radio receiver. This is not the limiting factor. In any receiving station the limiting factor is noise - weak signals are not limited by the actual signal level, but by the noise masks them out. This noise can come from a variety of sources. It can be picked up by the antenna or it can be generated within the radio receiver.

It is found that the level of noise that is picked up externally by a receiver from the antenna falls as the frequency increases. At HF and frequencies below this the combination of galactic, atmospheric and man-made noise is relatively high and this means that there is little point in making a receiver particularly sensitive. Normally radio receivers are designed such that the internally generated noise is much lower than any received noise, even for the quietest locations.

At frequencies above 30 MHz the levels of noise start to reach a point where the noise generated within the radio receiver becomes far more important. By improving the noise performance of the radio receiver, it becomes possible to detect much weaker signals.

Design for noise performanceIn terms of the receiver noise performance it is always the first stages or front end that is most crucial. At the front end the signal levels are at their lowest and even very small amounts of noise can be comparable with the incoming signal. At later stages in the set the signal will have been amplified and will be much larger. The same levels of noise as are present at the front end will be a much smaller proportion of the signal and will not have the same effect. Accordingly it is important that the noise performance of the front end is optimised for its noise performance.

It is for this reason that the noise performance of the first radio frequency amplifier within the receiver is of great importance. It is the performance of this circuit that is crucial in determining the performance of the whole radio receiver. To achieve the optimum performance for the first stage of the radio receiver there are a number of steps that can be taken. These include:

Determine the circuit topology required Choose a low noise device Determine the gain required Determine the current through the device Use low noise resistors Optimise the matching Ensure that power supply noise entering the circuit is removed

Determination of circuit topology   The first step in any design is to decide upon the type of circuit to be used. Whether a conventional common emitter style circuit is to be used, or even whether a common base should be employed. The decision will depend upon factors including the matching input and output impedances, the level of gain required and the matching arrangements to be used.

Choice of active device   The type of device to be used is also important. There are generally two decisions, whether to use a bipolar based transistor, or whether to use a field effect device. Having made this, it is obviously necessary to decide upon a low noise device. The noise performance of transistors and FETs is normally specified, and special high performance low noise devices are available for these applications.

Determination of required gain   While it may appear that the maximum level of gain may be required from this stage to minimise the levels of amplification required later and in this way ensure that the noise performance is optimised, this is not always the case. There are two major reasons for this. The first is that the noise performance of the circuit may be impaired by requiring too high a level of gain. Secondly it may lead to overload in later stages of the radio receiver and this may degrade the overall performance. Thus the level of gain required must be determined from the fact that it is necessary to optimise

the noise performance of this stage, and secondly to ensure that later stages of the receiver are not overloaded.

Determination of current through the active device   The design of the first stage of the radio receiver must be undertaken with care. To obtain the required RF performance in terms of bandwidth and gain, it may be necessary to run the device with a relatively high level of current. This will not always be conducive to obtaining the optimum noise performance. Accordingly the design must be carefully optimised to ensure the best performance for the whole radio receiver.

Use of low noise resistors   It may appear to be an obvious statement, but apart from choosing a low noise active device, consideration should also be given to the other components in the circuit. The other chief contributors are the resistors. The metal oxide film resistors used these days, including most surface mount resistors normally offer good performance in this respect and can be used as required.

Optimise impedance matching   In order to obtain the best noise performance for the whole radio receiver it is necessary to optimise the impedance matching. It may be thought that it is necessary to obtain a perfect impedance match. Unfortunately the best noise performance does not usually coincide with the optimum impedance match Accordingly during the design of the RF amplifier it is necessary to undertake some design optimisation to ensure the best overall performance is achieved for the radio receiver.

Ensure that power supply noise entering the circuit is removed   Power supplies can generate noise. In view of this it is necessary to ensure that any noise generated by the radio receiver power supply does not enter the RF stage. This can be achieved by ensuring that there is adequate filtering on the supply line to the RF amplifier.

SummaryReceiver sensitivity is one of the vital specifications of any radio receiver. The key factor in determining the sensitivity performance of the whole receiver is the RF amplifier. By optimising its performance, the figures for the whole of the receiver can be improved. In this way the specifications for signal to noise ratio, SINAD or noise figure can be brought to the required level.

 There are a number of ways in which the noise performance, and hence the sensitivity of a radio receiver can be measured. The most obvious method is to compare the signal and noise levels for a known signal level, i.e. the signal to noise (S/N) ratio or SNR. Obviously the greater the difference between the signal and the unwanted noise, i.e. the greater the S/N ratio, the better the radio receiver sensitivity performance.

As with any sensitivity measurement, the performance of the overall radio receiver is determined by the performance of the front end RF amplifier stage. Any noise introduced by the first RF amplifier will be added to the signal and amplified by subsequent amplifiers in the receiver. As the noise introduced by the first RF amplifier will be amplified the most, this RF amplifier becomes the most critical in terms of radio receiver sensitivity performance. Thus the first amplifier of any radio receiver should be a low noise amplifier.

Methods of measuring receiver sensitivityAlthough there are many ways of measuring the sensitivity performance of a radio receiver, the S/N ratio or SNR is one of the most straightforward and it is used in a variety of applications. However it has a number of limitations, and although it is widely used, other methods including noise figure are often used as well. Nevertheless the S/N ratio or SNR is an important specification, and it will be seen in many radio receiver specification sheets.

Signal to noise ratio for a radio receiver

The difference is normally shown as a ratio between the signal and the noise (S/N) and it is normally expressed in decibels. As the signal input level obviously has an effect on this ratio, the input signal level must be given. This is usually expressed in microvolts. Typically a certain input level required to give a 10 dB signal to noise ratio is specified.

Effect of bandwidthA number of other factors apart from the basic performance of the set can affect the SNR specification. The first is the actual bandwidth of the receiver. As the noise spreads out over all frequencies it is found that the wider the bandwidth of the receiver, the greater the level of the noise. Accordingly the receiver bandwidth needs to be stated.

Additionally it is found that when using AM the level of modulation has an effect. The greater the level of modulation, the higher the audio output from the receiver. When measuring the noise performance the audio output from the receiver is measured and accordingly the modulation level of the AM has an effect. Usually a modulation level of 30% is chosen for this measurement.

Typical figuresThis method of measuring the performance is most commonly used for HF communications receivers. Typically one might expect to see a figure in the region of 0.5 microvolts for a 10 dB S/N in a 3 kHz bandwidth for SSB or Morse. For AM a figure of 1.5 microvolts for a 10 dB S/N in a 6 kHz bandwidth at 30% modulation for AM might be seen.

Points to note when measuring SNRSNR is a very convenient method of quantifying the sensitivity of a receiver, but there are some points to note when measuring and interpreting the figures. To investigate these it is necessary to look at the way the measurements of SNR are made. A calibrated RF signal generator is used as a signal source for the receiver. It must have an accurate method of setting the output level down to very low signal levels. Then at the output of the receiver a true RMS AC voltmeter is used to measure the output level.

S/N and (S+N)/N   With the generator signal switched off a 50 Ohm match is given to the receiver and the audio meter will detect the noise generated by the receiver itself. This level is noted and the signal turned on. Its level is adjusted until the audio level meter reads a level which is 10 dB higher than just the noise on its own. The level of the generator is that required to give the 10 dB signal to noise ratio.

The last statement was not strictly true. Whilst the first reading of the noise is quite accurate, the second reading of the signal also includes some noise as well. In view of this many manufacturers will specify a slightly different ratio: namely signal plus noise to noise (S+N/N). In practice the difference is not particularly large, but the S+N/N ratio is more correct.

PD and EMF   Occasionally the signal generator level in the specification will mention that it is either PD or EMF. This is actually very important because there is a factor of 2:1 between the two levels. For example 1 microvolt EMF. and 0.5 microvolt PD are the same. The EMF (electro-motive force) is the open circuit

voltage, whereas the PD (potential difference) is measured when the generator is loaded. As a result of the way in which the generator level circuitry works it assumes that a correct (50 Ohm) load has been applied. If the load is not this value then there will be an error. Despite this most equipment will assume values in PD unless otherwise stated.

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Radio receiver SINAD measurement

- an overview of the SINAD measurement used in specifying the sensitivity performance of many radio receivers.

One of the measurements that can be made to assess and specify the sensitivity performance of a radio receiver is SINAD. While not used as widely as the signal to noise ratio, or noise figure it is nevertheless used commonly and can be found in the specifications of many radio receivers. SINAD is often used in conjunction with FM receivers, but it can also be used for AM and SSB quite easily.

As with any radio receiver, the design of the RF amplifier is key to its sensitivity performance. A poorly performing RF amplifier will degrade the performance of the whole receiver. However a high performance low noise RF amplifier will enable the overall set to provide a high level of sensitivity. Accordingly time should be focussed in the design of the RF amplifier in order that it should reach the required level of performance.

What is SINAD?SINAD is a measurement that can be used for any communication device to look at the degradation of the signal by unwanted or extraneous signals including noise and distortion. However the SINAD measurement is most widely used for measuring and specifying the sensitivity of a radio receiver.

The actual definition of SINAD is quite straightforward. It can be summarised as the ratio of the total signal power level (Signal + Noise + Distortion) to unwanted signal power (Noise + Distortion). Accordingly, the higher the figure for SINAD, the better the quality of the audio signal.

The SINAD figure is expressed in decibels (dB) and can be determined from the simple formula:

SINAD     =     10Log ( SND / ND )

where:SND = combined Signal + Noise + Distortion power levelND = combined Noise + Distortion power level

It is worth noting that SINAD is a power ratio and not a voltage ratio for this calculation.

Making SINAD measurementsTo make the measurement a signal modulated with an audio tone is entered into the radio receiver. A frequency of 1 kHz is taken as the standard as it falls in the middle of the audio bandwidth. A measurement of the whole signal, i.e. the signal plus noise plus distortion is made. As the frequency of the tone is known, the regenerated audio is passed through a notch filter to remove the tone. The remaining noise and distortion is then measured.

Although it is most common to measure the electrical output at the receiver audio output terminals, another approach that is not as widely used, is to pass the signal into the loudspeaker and then use a transducer connected to SINAD meter to convert the audio back into an electrical signal. This will ensure that any distortion included by the speaker

is incorporated, and it may overcome problems with gaining access to the speaker connections in certain circumstances where this may not be possible.

Obtaining the figures for the signal plus noise plus distortion and the noise plus distortion it is then possible to calculate the value of SINAD for the radio receiver of other piece of equipment.

The set up used for making SINAD measurements

While the measurements for SINAD can be made using individual items of test equipment, a number of SINAD meters are made commercially. These SINAD meters incorporate all the required circuitry and can be connected directly to radio receivers to make the measurements. Accordingly SINAD meters are a particularly convenient method of making these measurements.

Filter for SINAD measurementsThe notch filter that is required for SINAD measurements to be made has an effect on the measurement. In an ideal world the filter would be infinitely sharp a notch out only the modulating tone. However in the real world the filter will have a finite bandwidth. As its bandwidth increases, so it will remove noise and distortion as well as the tone. However as the distortion products will typically result from the second and third harmonics of the tone, the filter will not have an effect on this element of the reading. Nevertheless it may still have an effect on the noise levels.

In view of this problem some standards set down specifications or guidelines for the filter used in the SINAD measurement. ETSI (European Telecommunications Standards Institute) defines a notch filter (ETR 027). With the standard tone frequency of 1 kHz, it states that a filter used for SINAD measurements shall be such that the output the 1000 Hz tone shall be attenuated by at least 40 dB and at 2000 Hz the attenuation shall not exceed 0.6 dB. The filter characteristic shall be flat within 0.6 dB over the ranges 20 Hz to 500 Hz and 2000 Hz to 4000 Hz. In the absence of modulation the filter shall not cause more than 1 dB attenuation of the total noise power of the audio frequency output of the receiver under test.

In addition to the filter performance another critical area of a SINAD measurement is the measurement of the output signal power levels. These have to be a true power measurements that accommodate the different form factors of the variety of waveforms, i.e. sine wave for the 1 kHz tone and its harmonics, but the noise will be random and have a different form factor.

Applications of SINAD measurementsSINAD measurements give an assessment of the signal quality from a receiver under a number of conditions. As such SINAD measurements can be used for assessing a number of elements of receiver performance.

Receiver sensitivity:   The most common use of the SINAD measurement is to assess the sensitivity performance of a radio receiver. To achieve this the sensitivity can be assessed by determining the RF input level at the antenna that is required to achieve a given figure of SINAD. Normally a SINAD value of 12 dB is taken as this corresponds to a distortion factor of 25%, and a modulating tone of 1 kHz is used. It is also necessary to determine other conditions. For AM it is necessary to specify the depth of modulation and for FM

the level of deviation is required. For FM analogue systems ETSI specifies the use of a deviation level of 12.5% of the channel spacing

A typical specification might be that a receiver has a sensitivity of 0.25 uV [microvolts] for a 12 dB SINAD. Obviously the lower the input voltage needed to achieve the given level of SINAD, the better the receiver performance.

Adjacent channel rejection:   This parameter is a measure of the ability of the receiver to reject signals on a nearby channel. As the adjacent channel performance degrades, so the levels of noise and extraneous signals will increase, thereby degrading the SINAD performance.

An initial measurement of SINAD is made at a given level and this is known as the reference sensitivity. The RF input level of the signal for the SINAD measurement is then increased by 3 dB at the receiver antenna input. A second source or signal with modulated with a 400 Hz tone is added with its frequency set to an adjacent channel or at a specific offset from the carrier source used for the basic SINAD measurement. It will be found that the interferer will cause the 400 Hz tone to appear in the audio of the receiver as its level is increased. This will be seen as a degradation in the SINAD as the 400 Hz tone will pass through the SINAD meter notch filter.

With the measurement system set up, the interferer signal level is raised until the SINAD value is degraded to the original value obtained at the reference sensitivity. Then the ratio of the interfering level to the wanted signal is the adjacent channel rejection.

Receiver blocking:   SINAD can be used to form the basis of a receiver blocking measurement. As with other similar measurements a reference SINAD sensitivity level is found. The level of the SINAD signal is increased by 3 dB at the antenna. An un-modulated off channel signal is then added and its level raised until the receiver desensitises to an extent whereby the reference SINAD level is reached.

SummarySINAD is a particularly useful measurement format that can be used to determine the performance of a radio receiver under a variety of conditions. Although SINAD is primarily used to specify the basic sensitivity performance of many radios, it can be used for other parameters as well. Additionally it is chiefly used for FM systems, but its use is equally applicable to AM and SSB. It may also be used for digital systems as well, although this is not common practice as a measurement known as bit error rate (BER) is more widely used.

The overall figure for SINAD will be chiefly dependent upon the performance of the RF amplifier in the receiver. A low noise RF amplifier will enable the set as a whole to provide a good SINAD performance.

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Radio receiver noise figure

- an overview of noise figure used in specifying the sensitivity performance of radio receivers and their components.

Although there are a number of methods of determining the sensitivity of radio receivers and their associated elements, the noise figure is one of the most widely used methods. Not only is it widely used to assess the sensitivity performance or receivers, but it can be applied to complete receiving systems or to elements such as RF amplifiers. Thus it is possible to use the same notation to measure the noise performance of a whole receiver, or an RF amplifier. This makes it possible to determine whether a low noise amplifier may be suitable for a particular system by judging their relative levels of performance.

BasicsEssentially the measurement assesses the amount of noise each part of the system or the system as a whole introduces. This could be the radio receiver, or an RF amplifier for example. If the system were perfect then no noise would be added to the signal when it passed through the system and the signal to noise ratio would be the same at the output as at the input. As we all know this is not the case and some noise is always added. This means that the signal to noise ratio or SNR at the output is worse than the signal to noise ratio at the input. In fact the noise figure is simply the comparison of the SNR at the input and the output of the circuit.

A figure known as the noise factor can be derived simply by taking the SNR at the input and dividing it by the SNR at the output. As the SNR at the output will always be worse, i.e. lower, this means that the noise factor is always greater than one.

The noise factor is rarely seen in specifications. Instead the noise figure is always seen. This is simply the noise factor expressed in decibels.

Noise figure

In the diagram S1 is the signal at the input, N1 is the noise at the inputand S2 is the signal at the output and N2 the noise at the output

As an example if the signal to noise ratio at the input was 4:1, and it was 3:1 at the output then this would give a noise factor of 4/3 and a noise figure of 10 log (4/3) or 1.25 dB. Alternatively if the signal to noise ratios are expressed in decibels then it is quite easy to calculate the noise figure simply by subtracting one from another because two numbers are divided by subtracting their logarithms. In other words if the signal to noise ratio was 13 dB at the input and only 11 dB at the output then the circuit would have a noise figure of 13 - 11 or 2 dB.

Typical examplesThe specifications of different pieces of equipment will vary quite widely. A typical HF receiver may have a noise figure of 15 dB of more and function quite satisfactorily. A better level of performance is not necessary because of the high level of atmospheric noise. However an amateur receiver used on Two metres, for example, might have a noise figure of 3 or 4 dB. RF amplifiers for this band often have a noise figure of around 1 dB. However it is interesting to note that even the best professional wide-band VHF UHF receivers may only have a noise figure of around 8 dB.

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Radio receiver noise floor

- an overview of the noise floor or a receiver, what it is and how the noise floor affects the performance of a receiver.

Noise is a fact of life. Despite the best efforts of any design engineers, there is always some background noise present in any radio receiver. The noise emanates from many sources, and although the design of the receiver is optimised to reduce it some will always be present.

Accordingly a concept that is very useful in many elements of signal theory and hence in radio receiver design is that of a noise floor. The noise floor can be defined as the measure of the signal created from the sum of all the noise sources and unwanted signals within a system.

In order to reduce the levels of noise and thereby improve the sensitivity of the receiver, the main element of the receiver that requires its performance to be optimised is the RF amplifier. The use of a low noise amplifier at the front end of the receiver will ensure that its performance will be maximised. Wither for use at microwaves or lower frequencies, this RF amplifier is the chief element in determining the performance of the whole receiver. The next most important element is the first mixer.

Receiver noise floorWhile noise can emanate from many sources, when looking purely at the receiver, the noise is dependent upon a number of elements. The first is the minimum equivalent input noise for the receiver. This can be calculated from the following formula:

 

P     =     k T B

Where:P is the power in wattsK is Boltzmann's constant (1.38 x 10^-23 J/K)B is the bandwidth in Hertz

 

Using this formula it is possible to determine that the minimum equivalent input noise for a receiver at room temperature (290K) is -174 dBm / Hz.

It is then possible to calculate the noise floor for the receiver:

 

Noise floor     =     -174   +   NF +   10 log Bandwidth

Where NF is the noise figuredBm is the power level expressed in decibels relative to one milliwatt

 

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Radio receiver strong signal response- including intermodulation distortion, third order intercept point, cross modulation and blocking

Receiver sensitivity is important but equally so is the way in which a receiver handles strong signals. Specifications including intermodulation distortion, third order intercept point, cross modulation and blocking can be equally vital. In any receiver design a good balance must be achieved between the sensitivity and the strong signal handling capability. Under some conditions receivers may need to contend with signals that are only a few microvolts, but equally they need to handle the conditions when many millivolts enter the front end.,/p>

RF amplifierUnder normal conditions the RF amplifiers should remain linear with the output remaining proportional to the input. Unfortunately even the best amplifiers have limits to

their output capability, and beyond this they start to overload. When this happens their output starts to limit and the output is less than expected. At this point the amplifier is said to be in compression.

The characteristic curve for an amplifier

Compression in itself is not a problem. The absolute values of a signal are of little value and in any case the automatic gain control (AGC) used in most receivers means that the gain is reduced when strong signals are being received. However the side effects of compression give rise to major problems. Effects like intermodulation distortion, cross modulation, blocking and others mean that the operation of the receiver can be seriously impaired. It is these aspects which are of great importance in the receiver design.

To help prevent these problems occurring, receivers have a number of methods of reducing the signals levels. The most important is the AGC. This is standard on virtually every receiver and operates on many of the amplifier stages within the set. It prevents the signals from becoming too large, especially in the later stages of the set. However it cannot always prevent the front end stages from being overloaded. This is particularly true when the offending strong signal is slightly off channel. In this case it will enter the early stages of the set but not pass through the IF filters (assuming the receiver is a superhet). This will mean that the AGC will not be affected but the signal is still able to overload some of the early stages.

Some HF communications receivers have an attenuator on the input, although many receivers used in applications such as cellular telecommunications, PMR and the like will not have these and the receiver will need to be able to handle the strong signals without this assistance.

In view of the importance of the various aspects of overloading, a number of specifications quantify the various problems caused. However to look at these it is necessary to look at the effects and how they arise.

DistortionThe problems from compression arise as a result of the distortion which occurs to the signal when the amplifier runs into compression. The actual method which gives rise to problems may not be obvious at first sight. It can be viewed as the combination of two effects. However to see how it arises it is necessary to look at some of the basic effects of compression.

One of the forms of distortion which arises is harmonic distortion where harmonics of the wanted signal are produced. Depending upon the exact way in which the signal is compressed the levels of even order harmonics (2f, 4f, 6f, etc) and odd order harmonics (3f, 5f, 7f, etc) will vary. As a result of the production of these harmonics it is possible that signals below that being received could be picked up. However the RF selectivity is likely to remove these signals before they enter the first stages of the receiver.

Another effect which can be noticed is that the amplifier tends to act as a mixer. The non-linear transfer curve means that signals will mix together or modulate one another. This effect is known as intermodulation. It is unlikely that this effect on its own would give any problems. The mix products from signals close to the wanted one fall well away from

the received signal. Alternatively, to produce a signal within the receiver pass-band, signals well away from the received one would need to be entering the r.f. amplifier. These would normally be rejected by the RF selectivity. Take the example of two signals on 50.00 and 50.01 MHz. These would mix together to give signals at 0.01 MHz and 100.01 MHz. These are not likely to give rise to any problems.

Problems start to arise when the two effects combine with one another. It is quite possible for a harmonic of one signal to mix with the fundamental or a harmonic of the other. The third order sum products like 2f1 + f2 are unlikely to cause a problem, but the difference products like 2f1 - f2 can give significant problems. Take the example of a receiver set to 50 MHz where two strong signals are present, one at 50.00 MHz and the other at 50.01 MHz. The difference signals produced will be at 2 x 50.00 - 50.01 = 49.99 MHz and another at 2 x 50.01 - 50 = 50.01 MHz. As it can be seen either of these could cause interference on the band. Other higher order products can also cause problems: 3f1 - 2f2, 4f1 - 3f2, 5f1 - 4f2, and so forth all give products which may could pass through the receiver if it is tuned to the relevant frequency.

Intermodulation products from two signals

In this way the presence of a strong signal can produce other spurious signals which can appear in its vicinity. The signals mixing with one another in this way may be of a variety of different types, e.g. AM, FM, digital modulation, etc, all of which may combine together to give what is effectively noise. This means that poor third order intermodulation performance can have the effect of raising the noise floor under real operating conditions.

Third Order InterceptIt is found that the level of intermodulation products rise very fast. For a 1 dB increase in wanted signal levels, third order products will rise by 3 dB, and fifth order ones by 5 dB. This can be plotted to give a graph of the performance of the amplifier. Eventually the amplifier will run into saturation and the levels of all the signals will be limited. However if the curve of the wanted signals and the third order products was continued, the two lines would intersect. This is known as the third order intercept point. Naturally the higher the level of the intercept point, the better the performance of the amplifier. For a good receiver and intercept point of 25 dBm (i.e. 25 dB above 1 milliwatt or about 0.5 watt) might be expected.

The third order intercept point of an amplifier

BlockingWhen a very strong off channel signal appears at the input to a receiver it is often found that the sensitivity is reduced. The effect arises because the front end amplifiers run into compression as a result of the off channel signal. This often arises when a receiver and transmitter are run from the same site and the transmitter signal is exceedingly strong. When this occurs it has the effect of suppressing all the other signals trying to pass through the amplifier, giving the effect of a reduction in gain.

Blocking is generally specified as the level of the unwanted signal at a given offset (normally 20 kHz) which will give a 3 dB reduction in gain. A good receiver may be able to withstand signals of about ten milliwatts before this happens.

Cross modulationAnother effect which can be noticed when there are strong signals entering the receiver is known as cross modulation. When this occurs the modulation from a strong signal can be transferred onto other signals being picked up. This effect is particularly obvious when amplitude modulated signals are being received. In this case the modulation of another signal can be clearly heard.

Cross modulation normally arises out of imperfect mixer performance in the radio, although it can easily occur in one of the RF amplifiers. As it is a third order effect, a receiver with a good third order intercept point should also exhibit good cross modulation performance.

To specify the cross modulation performance the effect of a strong AM carrier on a smaller wanted signal is noted. Generally the level of a strong carrier with 30% modulation needed to produce an output 20 dB below that produced by the wanted signal. The wanted signal level also has to be specified and 1mV or -47dBm (i.e. a signal 47 dB below 1 mW) is often taken as standard, together with an offset frequency of 20 kHz.

 Sensitivity is one of the main specifications of any radio receiver. However the sensitivity of a set is by no means the whole story. The specification for a set may show it to have an exceedingly good level of sensitivity, but when it is connected to an antenna its performance may be very disappointing because it is easily overloaded when strong signals are present, and this may impair its ability to receive weak signals.

The overall dynamic range of the receiver is very important. It is just as important for a set to be able to handle strong signals well as it is to be able to pick up weak ones. This becomes very important when trying to pick up weak signals in the presence of nearby strong ones. Under these circumstances a set with a poor dynamic range may not be able to hear the weak stations picked up by a less sensitive set with a better dynamic range. Problems like blocking, inter-modulation distortion and the like within the receiver may mask out the weak signals, despite the set having a very good level of sensitivity.

What is dynamic range?The dynamic range of a receiver is essentially the range of signal levels over which it can operate. The low end of the range is governed by its sensitivity whilst at the high end it is governed by its overload or strong signal handling performance. Specifications generally use figures based on either the inter-modulation performance or the blocking performance. Unfortunately it is not always possible to compare one set with another because dynamic range like many other parameters can be quoted in a number of ways. However to gain an idea of exactly what the dynamic range of a receiver means it is worth looking at the ways in which the measurements are made to determine the range of the receiver.

SensitivityThe first specification to investigate is the sensitivity of a set. The main limiting factor in any receiver is the noise generated. For most applications either the signal to noise ratio or the noise figure is used as described in a previous issue of MT. However for dynamic range specifications a figure called the minimum discernible signal (MDS) is often used.

This is normally taken as a signal equal in strength to the noise level. As the noise level is dependent upon the bandwidth used, this also has to be mentioned in the specification. Normally the level of the level of the MDS is given in dBm i.e. dB relative to a milliwatt and typical values are around -135 dBm in a 3 kHz bandwidth.

Strong signal handlingAlthough the sensitivity is important the way in which a receiver handles strong signals is also very important. Here the overload performance governs how well the receiver performance.

In the ideal world the output of an amplifier would be proportional to the input for all signal levels. However amplifiers only have a limited output capability and it is found that beyond a certain level the output falls below the required level because it cannot handle the large levels required of it. This gives a characteristic like that shown in Fig. 1. From this it can be seen that amplifiers are linear for the lower part of the characteristic, but as the output stages are unable to handle the higher power levels the signals starts to become compressed as seen by the curve in the characteristic.

A typical amplifier characteristic

The fact that the amplifier is non-linear does not create a major problem in itself. However the side effects do. When a signal is passed through a non-linear element there are two main effects which are noticed. The first is that harmonics are generated. Fortunately these are unlikely to cause a major problem. For a harmonic to fall near the frequency being received, a signal at half the received frequency must enter the amplifier. The front end tuning should reduce this by a sufficient degree for it not to be a noticeable problem under most circumstances.

The other problem that can be noticed is that signals mix together to form unwanted products. These again are unlikely to cause a problem because any signals which could mix together should be removed sufficiently by the front end tuning. Instead problems occur when harmonics of in-band signals mix together.

Third order productsProblems occur when harmonics of in-band signals mix together. It is found that a comb of signals can be produced as shown in Figure 2, and these may just fall on the same frequency as a weak and intersting station, thereby masking it out so it cannot be heard.

It is simple to calculate the frequencies where the spurious signals will fall. If the input frequencies are f1 and f2, then the new frequencies produced will be at 2f1 - f2, 3f1 - 2f2, 4f1

- 3f2 and so forth. On the other side of the two main or original signals products are produced at 2f2 - f1, 3f2 - 2f2, 4f2 - 3f1 and so forth as shown in the diagram. These are known as odd order inter-modulation products. Two times one signal plus one times another makes a third order product, three times one plus two times another is a fifth order product and so forth. It can be seen from the diagram that the signals either side of the main signals are first the third order product, then fifth, seventh and so forth.

To take an example with some real figures. If large signals appear at frequencies of 30.0 MHz and 30.01 MHz, then the inter-modulation products will appear at 30.02, 30.03, 30.4 ...MHz and 29.99, 29.98, 29.97 ..... MHz.

Inter-modulation products

BlockingAnother problem that can occur when a strong signal is present is known as blocking. As the name implies it is possible for a strong signal to block or at least reduce the sensitivity of a receiver. The effect can be noticed when listening to a relatively weak station and a nearby transmitter starts to radiate, and the wanted signal reduces in strength. The effect is caused when the front-end amplifier starts to run into compression. When this occurs the strongest signal tends to "capture" the amplifier reducing the strength of the other signals. The effect is the same as the capture effect associated with FM signals.

The amount of blocking is obviously dependent upon the level of the signal. It also depends on how far off channel the strong signal is. The further away, the more it will be reduced by the front end tuning and the less the effect will be. Normally blocking is quoted as the level of the unwanted signal at a given offset (normally 20 kHz) to give a 3 dB reduction in gain.

Dynamic range definitionWhen looking at dynamic range specifications, care must be taken when interpreting them. The MDS at the low signal end should be viewed carefully, but the limiting factors at the top end show a much greater variation tin the way they are specified. Where blocking is used a reduction of 3 dB sensitivity is normally specified, but in some cases may be 1 dB used. Where the inter-modulation products are chosen as the limiting point the input signal level for them to be the same as the MDS is often taken. However whatever specification is given, care should be taken to interpret the figures as they may be subtlety different in the way they are measured from one receiver to the next.

To gain a feel for the figures which may be obtained where inter-modulation is the limiting factor figures of between 80 and 90 dB range are typical, and where blocking is the limiting factor figures around 115 dB are generally achieved in a good receiver.

Designing for optimum performanceIt is not an easy task to design a highly sensitive receiver that also has a wide dynamic range. To achieve this performance a number of methods can be used. The front-end stage is the most critical in terms of noise performance. It should be optimised for noise performance rather than gain. Input impedance matching is critical for this. It is interesting to note that the optimum match does not correspond exactly with the best noise performance. The amplifier should also have a relatively high output capability to ensure it does not overload. The mixer is also critical to the overload performance. To ensure the mixer is not overloaded there should not be excessive gain preceding it. A high level mixer should also be used (i.e. one designed to accept a high-level local oscillator signal). In this way it can tolerate high input signals without degradation in performance. Care should be taken in the later stages of the receiver to ensure that they can tolerate the level of signals likely to be encountered. A good AGC system also helps prevent overloading and the generation of unwanted spurious signals.

A receiver with a good dynamic range will be able to give a far better account of itself under exacting conditions than one designed purely for optimum sensitivity.

Frequency modulation is widely used in radio communications and broadcasting, particularly on frequencies above 30 MHz. It offers many advantages, particularly in mobile radio applications where its resistance to fading and interference is a great advantage. It is also widely used for broadcasting on VHF frequencies where it is able to provide a medium for high quality audio transmissions.

In view of its widespread use, a wide variety of receivers are able to demodulate these transmissions. Naturally there are specifications and figures that receiver manufacturers quote for the performance of their sets when receiving FM. These include sch figures as quieting, capture ratio and the like.

Receiving FMIn order to be able to receive FM a receiver must be sensitive to the frequency variations of the incoming signals which may be wide or narrow band. However the set is made insensitive to the amplitude variations. This is achieved by having a high gain IF amplifier. Here the signals are amplified to such a degree that the amplifier runs into limiting. In this way any amplitude variations are removed and this improves the signal to noise ratio after the point when the signal limits in the IF stages. However the high levels of gain associated with the limiting process mean that when no signal is present, very high levels of noise appear at the output of the FM demodulator.,/p>

SquelchTo overcome the problem of the high noise levels when no signal is present a circuit known as "squelch" is normally used. This detects when no signal is present and cuts the audio, thereby removing the noise under these conditions. The level for this is normally present in domestic radios, but there is often a level adjustment for PMR or handheld transceivers, or for scanners and professional receivers.

Quieting specificationOne of the advantages of FM is its resilience to noise. This is one of the main reasons why it is used for high quality audio broadcasts. However when no signal is present, a high noise level is present at the output of the receiver. If a low level FM signal is introduced and its level slowly increased it will be found that the noise level reduces. From this the quieting level can be deduced. It is the reduction in noise level expressed in decibels when a signal of a given strength is introduced to the input of the set. Typically a broadcast tuner should give a quieting level of 30 dB for an input level of around a microvolt.

Capture effectAnother effect that is often associated with FM is called the capture effect. This can be demonstrated when two signals are present on the same frequency. When this occurs it is found that only the stronger signal will heard at the output This can be compared to AM where a mixture of the two signals is heard, along with a heterodyne if there is a frequency difference.

A capture ratio is often defined in receiver specifications. It is the ratio between the wanted and unwanted signal to give a certain reduction in level of the unwanted signal at the output. Normally a reduction of the unwanted signal of 30 dB is used. To give an example of this the capture ratio may be 2 dB for a typical tuner to give a reduction of 30 dB in the unwanted signal. In other words if the wanted signal is only 2 dB stronger than the unwanted one, the audio level of the unwanted one will be suppressed by 30 dB.

 The phase locked loop or PLL is a particularly flexible circuit building block. The phase locked loop, PLL can be used for a variety of radio frequency applications, and accordingly the PLL is found in many radio receivers as well as other pieces of equipment.

The phase locked loop, PLL, was not used in early radio equipment because of the number of different stages required. However with the advent of radio frequency integrated circuits, the idea of phase locked loops, PLLs, became viable. Initially relatively low frequency PLLs became available, but as RF IC technology improved, so

the frequency at which PLLs would operate rose, and high frequency versions became available.

Phase locked loops are used ain a large variety of applications within radio frequency technology. PLLs can be used as FM demodulators and they also form the basis of indirect frequency synthesizers. In addition to this they can be used for a number of applications including the regeneration of chopped signals such as the colour burst signal on an analogue colour television signal, for types of variable frequency filter and a host of other specialist applications

Concepts - phaseThe operation of a phase locked loop, PLL, is based around the idea of comparing the phase of two signals. This information about the error in phase or the phase difference between the two signals is then used to control the frequency of the loop.

To understand more about the concept of phase and phase difference, first visualise a radio frequency signal in the form of a familiar x-y plot of a sine wave. As time progresses the amplitude oscillates above and below the line, repeating itself after each cycle. The linear plot can also be represented in the form of a circle. The beginning of the cycle can be represented as a particular point on the circle and as a time progresses the point on the waveform moves around the circle. Thus a complete cycle is equivalent to 360 degrees. The instantaneous position on the circle represents the phase at that given moment relative to the beginning of the cycle.

To look at the concept of phase difference, take the example of two signals. Although the two signals have the same frequency, the peaks and troughs do not occur in the same place. There is said to be a phase difference between the two signals. This phase difference is measured as the angle between them. It can be seen that it is the angle between the same point on the two waveforms. In this case a zero crossing point has been taken, but any point will suffice provided that it is the same on both.

When there two signals have different frequencies it is found that the phase difference between the two signals is always varying. The reason for this is that the time for each cycle is different and accordingly they are moving around the circle at different rates. It can be inferred from this that the definition of two signals having exactly the same frequency is that the phase difference between them is constant. There may be a phase difference between the two signals. This only means that they do not reach the same point on the waveform at the same time. If the phase difference is fixed it means that one is lagging behind or leading the other signal by the same amount, i.e. they are on the same frequency.

PLL basicsA phase locked loop, PLL, is basically of form of servo loop. Although a PLL performs its actions on a radio frequency signal, all the basic criteria for loop stability and other parameters are the same.

A basic phase locked loop, PLL, consists of three basic elements:

Phase comparator:   As the name implies, this circuit block within the PLL compares the phase of two signals and generates a voltage according to the phase difference between the two signals.

Loop filter:   This filter is used to filter the output from the phase comparator in the PLL. It is used to remove any components of the signals of which the phase is being compared from the VCO line. It also governs many of the characteristics of the loop and its stability.

Voltage controlled oscillator (VCO):   The voltage controlled oscillator is the circuit block that generates the output radio frequency signal. Its frequency can be controlled and swung over the operational frequency band for the loop.

PLL operationThe concept of the operation of the PLL is relatively simple, although the mathematical analysis can become more complicated

The Voltage Controlled Oscillator, VCO, within the PLL produces a signal which enters the phase detector. Here the phase of the signals from the VCO and the incoming reference signal are compared and a resulting difference or error voltage is produced. This corresponds to the phase difference between the two signals.

Block diagram of a basic phase locked loop (PLL)

The error signal from the phase detector in the PLL passes through a low pass filter which governs many of the properties of the loop and removes any high frequency elements on the signal. Once through the filter the error signal is applied to the control terminal of the VCO as its tuning voltage. The sense of any change in this voltage is such that it tries to reduce the phase difference and hence the frequency between the two signals. Initially the loop will be out of lock, and the error voltage will pull the frequency of the VCO towards that of the reference, until it cannot reduce the error any further and the loop is locked.

When the PLL is in lock a steady state error voltage is produced. By using an amplifier between the phase detector and the VCO, the actual error between the signals can be reduced to very small levels. However some voltage must always be present at the control terminal of the VCO as this is what puts onto the correct frequency.

The fact that a steady error voltage is present means that the phase difference between the reference signal and the VCO is not changing. As the phase between these two signals is not changing means that the two signals are on exactly the same frequency.

SummaryThe phase locked loop is one of the most versatile building blocks in radio frequency electronics today. Whilst it was not widely used for many years, the advent of the IC meant that phase locked loop and synthesizer chips became widely available. This made them cheap to use and their advantages could be exploited to the full. Nowadays most hi-fi tuners and car radios use them and a large proportion of the portable radios on the market as well. With their interface to microprocessors so easy their use is assured for many years to come.

The phase detector is the core element of a phase locked loop, PLL. Its action enables the phase differences in the loop to be detected and the resultant error voltage to be produced.

There is a variety of different circuits that can be used as phase detectors, some that use what may be considered as analogue techniques, while others use digital circuitry. However the most important difference is whether the phase detector is sensitive to just phase or whether it is sensitive to frequency and to phase. Thus phase detectors may be split into two categories:

Phase only sensitive detectors Phase - frequency detectors

Phase only sensitive detectorsPhase detectors that are only sensitive to phase are the most straightforward form of detector. They simply produce an output that is proportional to the phase difference between the two signals. When the phase difference between the two incoming signals is steady, they produce a constant voltage. When there is a frequency difference between the two signals, they produce a varying voltage. In fact the simplest form of phase only

sensitive detector is a mixer. From this it can be seen that the output signal will be have sum and difference signals.

The difference frequency product is the one used to give the phase difference. It is quite possible that the difference frequency signal will fall outside the pass-band of the loop filter. If this occurs then no error voltage will be fed back to the Voltage Controlled Oscillator (VCO) to bring it into lock. This means that there is a limited range over which the loop can be brought into lock, and this is called the capture range. Once in lock the loop can generally be pulled over a much wider frequency band.

To overcome this problem the oscillator must be steered close to the reference oscillator frequency. This can be achieved in a number of ways. One is to reduce the tuning range of the oscillator so that the difference product will always fall within the pass-band of the loop filter. In other instances another tune voltage can be combined with the feedback from the loop to ensure that the oscillator is in the correct region. This is approach is often adopted in microprocessor systems where the correct voltage can be calculated for any given circumstance.

Phase - frequency sensitive detectorsAnother form of detector is said to be phase-frequency sensitive. These circuits have the advantage that whilst the phase difference is between +/- 180 a voltage proportional to the phase difference is given. Beyond this the circuit limits at one of the extremes. In this way no AC component is produced when the loop is out of lock and the output from the phase detector can pass through the filter to bring the phase locked loop, PLL, into lock.

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Voltage controlled oscillator, VCO, for PLLs

- an overview of the various types of voltage controlled oscillator, VCO, used in phase locked loops, PLLs and frequency synthesizers

Within a phase locked loop, PLL, or frequency synthesizer, the performance of the voltage controlled oscillator, VCO is of paramount importance. In order that the PLL or synthesizer can meet its full specification a well designed oscillator is essential. Designing a really high performance voltage controlled oscillator, VCO, is not always easy as there are a number of requirements that need to be met. However by careful design, and some experimentation a good VCO design can be found.

VCO requirementsJust like any other circuit, with a VCO there are a number of design requirements that need to be known from the beginning of the design process. These basic requirements for the VCO will govern many of the decisions concerning the circuit topology and other fundamental aspects of the circuit. Some of the basic requirements are:<.p>

Tuning range

Tuning gain - tuning shift for a given tuning voltage change

Phase noise (low phase noise)

These are some of the main requirements that need to be known from the outset of the design of the VCO. The overall tuning range and the gain are basic requirements that are part of the basic design of any PLL into which the VCO may be incorporated. So too is

the phase noise characteristic. As phase noise is a basic parameter of any PLL or frequency synthesizer, so too is the characteristic of the VCO, and low phase noise VCOs are often required. For example the VCO performance may govern the overall design of the frequency synthesizer or PLL, if a given phase noise performance is to be met.

VCO circuitsLike any oscillator, a VCO may be considered as an amplifier and a feedback loop. The gain of the amplifier may be denoted as A and the feedback as B.

For the circuit to oscillate the total phase shift around the loop must be 360 degrees and the gain must be unity. In this way signals are fed back round the loop so that they are additive and as a result, any small disturbance in the loop is fed back and builds up. In view of the fact that the feedback network is frequency dependent, the build up of signal will occur on one frequency, the resonant frequency of the feedback network, and a single frequency signal is produced.

Many oscillators and hence VCOs use a common emitter circuit. This in itself produces a phase shift of 180 degrees, leaving the feedback network to provide a further 180 degrees.

Other oscillator or VCO circuits may use a common base circuit where there is no phase shift between the emitter and collector signals (assuming a bipolar transistor is used) and the phase shift network must provide either 0 degrees or 360 degrees.

Colpitts and Clapp VCO circuitsTwo commonly used examples of VCO circuits are the Colpitts and Clapp oscillators. Of the two, the Colpitts circuit is the most widely used, but these circuits are both very similar in their configuration.

These circuits operate as oscillators because it is found that a bipolar transistor with capacitors placed between the base and emitter (C1) and the emitter and ground (C2) fulfils the criteria required for providing sufficient feedback in the correct phase to produce an oscillator. For oscillation to take place the ratio C1: C2 must be greater than one.

The resonant circuit is made by including a inductive element between the base and ground. In the Colpitts circuit this consists of just an inductor, whereas in the Clapp circuit an indictor and capacitor in series are used.

The conditions for resonance is that:

f^2     =    1 / (4 pi^2 L C )

The capacitance for the overall resonant circuit is formed by the series combination of the two capacitors C1 and C2 in series. In the case of the Clapp oscillator, the capacitor in series with the inductor is also included in series with C1 and C2.

Thus the series capacitance is:

Ctot     =    1 / C1 + 1 / C2

In order to make the oscillator tune it is necessary to vary the resonant point of the circuit. This is best achieved by adding a capacitor across the indictor in the case of the Colpitts oscillator. Alternatively for the Clapp oscillator, it can be the capacitor in series with the inductor.

For high frequency applications a circuit where the inductive reactance is placed between the base and ground is often preferred as it is less prone to spurious oscillations and other anomalies.

Choice of VCO active deviceIt is possible to use both bipolar devices and FETs within a VCO, using the same basic

circuit topologies. The bipolar transistor has a low input impedance and is current driven, while the FET has a high input impedance and is voltage driven. The high input impedance of the FET is able to better maintain the Q of the tuned circuit and this should give a better level of performance in terms of the phase noise performance where the maintenance of the Q of the tuned circuit is a key factor in the reduction of phase noise.

Another major factor is the flicker noise generated by the devices. Oscillators are highly non-linear circuits and as a result the flicker noise is modulated onto VCO as sidebands and this manifests itself as phase noise. In general bipolar transistors offer a lower level of flicker noise and as a result VCOs based around them offer a superior phase noise performance.

VCO tuningTo make a VCO, the oscillator needs to be tuned by a voltage. This can be achieved by making the variable capacitor from varactor diodes. The tune voltage for the VCO can then be applied to the varactors.

When varactor diodes are used, care must be taken in the design of the circuit to ensure that the drive level in the tuned circuit is not too high. If this is the case, then the varactor diodes may be driven into forward conduction, reducing the Q and increasing the level of spurious signals.

There are two main types of varactor diode that may be used within a VCO: abrupt and hyper-abrupt diodes. The names refer to the junction within the diode. The abrupt ones do not have a sharp a transition between the two semiconductor types in the diode, and this affect the performance offered.

Hyper-abrupt diodes have a relatively linear voltage : capacitance curve and as a result they offer a very linear tuning characteristic that may be required in some applications. They are also able to tune over a wide range, and may typically tune over an octave range with less than a 20 volt change in tuning voltage. However they do not offer a particularly high level of Q. As this will subtract from the overall Q of the tuned circuit this will mean that the phase noise performance is not optimum.

Abrupt diodes, while not offering such a high tuning range or linear transfer characteristic are able to offer a higher Q. This results in a better phase noise (i.e. low phase noise) performance for the VCO. The other point to note is that they may need a high tuning voltage to provide the required tuning range, as some diodes may require a tuning voltage for the VCO to vary up to 50 volts or slightly more.

SummaryThe design of a VCO can be interesting and challenging. Whether the aim is to design a low noise VCO, a low current VCO, a PLL VCO, or one that will cover a wide tuning range there are many aspects that need to be addressed. Often when a successful design has been obtained, it will slightly modified to enable it to cover a wide range of similar applications.

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Oscillator design for low phase noise

- an overview of the design of radio frequency, RF, oscillators for low phase noise levels.

One of the key requirements for any oscillator used in a radio receiver, radio transmitter, or many other applications is for the oscillator to perform with low levels of phase noise. Whether the oscillator is used in a frequency synthesizer, or in any other application, the basic design principles for achieving low phase noise are the same.

Poor levels of oscillator can manifest themselves in slightly different ways. For an analogue radio receiver a poor performance oscillator may result in poor reciprocal mixing performance. It may also raise the noise floor of the receiver. In a radio system relying on phase modulation, phase noise will degrade the bit error rate performance. Additionally transmitters exhibiting a poor phase noise performance will tend to transmit wide band noise, causing interference to users on other frequencies.

Key points for oscillator designThere are some areas points to address when designing an oscillator to ensure that it has a good phase noise performance. By addressing these and other relevant points, the performance of the oscillator meets its requirements.

High Q resonant circuit

Choice of oscillator device

Correct feedback level

Sufficient oscillator power output

Power line rejection

Oscillator design methodologyIn order that the oscillator is able to provide the optimum phase noise performance it is necessary to implement these elements into the design of the circuit from the outset. Some aspects may affect the basic design criteria, and therefore need to be included from the concept stage of the circuit:

Q of the resonant circuit:    One of the major factors in determining the phase noise performance of an oscillator is the Q of the resonant circuit. Broadly, the higher the Q of the oscillator tuned circuit, the better the phase noise performance. This inductors should be chosen to provide the highest Q, as should the capacitors. This is particularly true of voltage controlled oscillators, VCOs where the varactor diodes normally employed have a lower Q than other capacitors.

Typically high Q tuned circuits do not have the tuning range of lower Q circuits. This means that when wide tuning ranges are required, it becomes more difficult to obtain a high level of Q and hence the optimum phase noise.

As an illustration of the effect of having a high Q resonant circuit in an oscillator, crystal oscillators exhibit very low levels of phase noise as a result of the fact that the crystals used in them possess very high levels of Q.

Choice of oscillator active device:    It is possible to use both bipolar devices and FETs within an RF oscillator, using the same basic circuit topologies. The bipolar transistor has a low input impedance and is current driven, while the FET has a high input impedance and is voltage driven. The high input impedance of the FET is able to better maintain the Q of the tuned circuit and this should give a better level of performance in terms of the phase noise performance where the maintenance of the Q of the tuned circuit is a key factor in the reduction of phase noise.

Another major factor is the flicker noise generated by the devices. Oscillators are highly non-linear circuits and as a result the flicker noise is modulated onto the oscillation as sidebands. This manifests itself as phase noise. In general bipolar transistors offer a lower level of flicker noise and as a result oscillators based around them offer a superior phase noise performance.

Oscillator feedback level:    A critical feature in any oscillator design is to ensure that the correct level of feedback is maintained. There should be sufficient to ensure that oscillation is maintained over the frequency range, over the envisaged temperature range and to accommodate the gain and parameter variations between the devices used. However if the level of feedback is too high, then the level of noise will also be increased. Thus the circuit should be designed to provide sufficient feedback for reliable operation and little more.

Sufficient oscillator power output:    It is found that the noise floor of an oscillator is reasonably constant in absolute terms despite the level of the output signal. In some designs there can be improvements in the overall signal to noise floor level to be made by using a high level signal and applying this directly to the mixer or other circuit where it may be required. Accordingly some low noise circuits may use surprisingly high oscillator power levels.

Power line rejection:    It is necessary to ensure that any supply line or other extraneous noise is not presented to the oscillator. Supply line ripple, or other unwanted pickup can seriously degrade the performance of the oscillator. To overcome this, good supply smoothing and regulation is absolutely necessary. Additionally it may be advisable to place the oscillator within a screened environment so that it does not pick up any stray noise. It is worth remembering that the oscillator acts as a high gain amplifier, especially close to the resonant frequency. Any noise picked up can be amplified and will manifest itself as phase noise.

SummaryThere are many elements to ensuring that an oscillator circuit design meets its requirements for low phase noise. The points provider here give a start to some of the basic decisions that are needed. Once initially realised, some refinement is likely to be needed to ensure the optimum performance is obtained.

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PLL loop filter

- an overview of the loop filter used in a phase locked loop, PLL. This gives an overview of the requirements, and design.

The design of the PLL, loop filter is crucial to the operation of the whole phase locked loop. The choice of the circuit values here is usually a very carefully balanced compromise between a number of conflicting requirements.

The PLL filter is needed to remove any unwanted high frequency components which might pass out of the phase detector and appear in the VCO tune line. They would then appear on the output of the Voltage Controlled Oscillator, VCO, as spurious signals. To show how this happens take the case when a mixer is used as a phase detector. When the loop is in lock the mixer will produce two signals: the sum and difference frequencies. As the two signals entering the phase detector have the same frequency the difference frequency is zero and a DC voltage is produced proportional to the phase difference as expected. The sum frequency is also produced and this will fall at a point equal to twice the frequency of the reference. If this signal is not attenuated it will reach the control voltage input to the VCO and give rise to spurious signals.

When other types of phase detector are used similar spurious signals can be produced and the filter is needed to remove them.

The filter also affects the ability of the loop to change frequencies quickly. If the filter has a very low cut-off frequency then the changes in tune voltage will only take place slowly, and the VCO will not be able to change its frequency as fast. This is because a filter with a low cut-off frequency will only let low frequencies through and these correspond to slow changes in voltage level.

Conversely a filter with a higher cut-off frequency will enable the changes to happen faster. However when using filters with high cut-off frequencies, care must be taken to ensure that unwanted frequencies are not passed along the tune line with the result that spurious signals are generated.

The loop filter also governs the stability of the loop. If the filter is not designed correctly then oscillations can build up around the loop, and large signals will appear on the tune line. This will result in the VCO being forced to sweep over wide bands of frequencies. The proper design of the filter will ensure that this cannot happen under any circumstances.

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PLL Frequency synthesizer tutorial- an introduction to the indirect (phase locked loop - pll) synthesizer

Today most receivers use a phase locked loop or PLL frequency synthesizer. Many of them advertise this fact by displaying words like "PLL", "Synthesized", or "Quartz" on their front panels or in the advertising literature. Whatever one thinks of the sales language, PLL frequency synthesizers offer tremendous advantages to the operation of a receiver. Not only do frequency synthesizers enable receivers to have the same stability as the quartz reference, but they also enable many other facilities to be introduced because they can easily be controlled by a microprocessor. This enables facilities such as multiple memories, keypad frequency entry, scanning and much more to be incorporated into the set.

Phase locked loop, PLL, frequency synthesizers are widely used, but their operation is not always well understood. One of the reasons for this is that their design can involve some complicated math, but despite this the basic concepts are relatively easy to grasp.

PLL BasicsA frequency synthesizer is based around a phase locked loop or PLL. This circuit uses the idea of phase comparison as the basis of its operation. From the block diagram of a basic loop shown in Fig. 1 it can be seen that there are three basic circuit blocks, a phase comparator, voltage controlled oscillator, and loop filter. A reference oscillator is sometimes included in the block diagram, although this is not strictly part of the loop itself even though a reference signal is required for its operation.

Block diagram of a basic phase locked loop (PLL)

The phase locked loop, PLL, operates by comparing the phase of two signals. The signals from the voltage controlled oscillator and reference enter the phase comparator Here a third signal equal to the phase difference between the two input signals is produced.

The phase difference signal is then passed through the loop filter. This performs a number of functions including the removal of any unwanted products that are present on this signal. Once this has been accomplished it is applied to the control terminal of the voltage controlled oscillator. This tune voltage or error voltage is such that it tries to reduce the error between the two signals entering the phase comparator. This means that the voltage controlled oscillator will be pulled towards the frequency of the reference, and when in lock there is a steady state error voltage. This is proportional to the phase error between the two signals, and it is constant. Only when the phase between two signals is changing is there a frequency difference. As the phase difference remains constant when the loop is in lock this means that the frequency of the voltage controlled oscillator is exactly the same as the reference.

SynthesisizersA phase locked loop, PLL, needs some additional circuitry if it is to be converted into a frequency synthesizer. This is done by adding a frequency divider between the voltage controlled oscillator and the phase comparator as shown in Fig. 2.

A programmable divider added into a phase locked loop, PLL, enables the frequency to be changed.

Programmable dividers or counters are used in many areas of electronics, including many radio frequency applications. They take in a pulse train like that shown in Fig. 3, and give out a slower train. In a divide by two circuit only one pulse is given out for every two that are fed in and so forth. Some are fixed, having only one division ratio. Others are programmable and digital or logic information can be fed into them to set the division ratio.

Operation of a programmable divider

When the divider is added into the circuit the phase locked loop, PLL, still tries to reduce the phase difference between the two signals entering the phase comparator. Again when the circuit is in lock both signals entering the comparator are exactly the same in frequency. For this to be true the voltage controlled oscillator must be running at a frequency equal to the phase comparison frequency times the division ratio.

It can be seen that if the division ratio is altered by one, then the voltage controlled oscillator will have to change to the next multiple of the reference frequency. This means that the step frequency of the synthesizer is equal to the frequency entering the comparator.

Most synthesizers need to be able to step in much smaller increments if they are to be of any use. This means that the comparison frequency must be reduced. This is usually

accomplished by running the reference oscillator at a frequency of a megahertz or so, and then dividing this signal down to the required frequency using a fixed divider. In this way a low comparison frequency can be achieved.

Comparison frequency reduced by adding a fixed divider after the reference oscillator

Analogue TechniquesPlacing a digital divider is not the only method of making a synthesizer using a phase locked loop, PLL. It is also possible to use a mixer in the loop as shown in Fig. 5. Using this technique places an offset into the frequency generated by the loop.

A phase locked loop, PLL, with mixer

The way in which the phase locked loop, PLL, operates with the mixer incorporated can be analyzed in the same manner that was used for the loop with a divider. When the loop is in lock the signals entering the phase detector are at exactly the same frequencies. The mixer adds an offset equal to the frequency of the signal entering the other port of the mixer. To illustrate the way this operates figures have been included. If the reference oscillator is operating at a frequency of 10 MHz and the external signal is at 15 MHz then the VCO must operate at either 5 MHz or 25 MHz.. Normally the loop is set up so that mixer changes the frequency down and if this is the case then the oscillator will be operating at 25 MHz.

It can be seen that there may be problems with the possibility of two mix products being able to give the correct phase comparison frequency. It happens that as a result of the phasing in the loop, only one will enable it to lock. However to prevent the loop getting into an unwanted state the range of the VCO is limited. For phase locked loops, PLLs, that need to operate over a wide range a steering voltage is added to the main tune voltage so that the frequency of the loop is steered into the correct region for required conditions. It is relatively easy to generate a steering voltage by using digital information from a microprocessor and converting this into an analogue voltage using a digital to

analogue converter (DAC). The fine tune voltage required to pull the loop into lock is provided by the loop in the normal way.

Multi-loop synthesizersMany high performance synthesizers use several loops that incorporate both mixers and digital dividers. By using these techniques it is possible to produce high performance wide range signal sources with very small step sizes. If only a single loop is used then there may be short falls in the level of performance.

There is a large variety of ways in which multi-loop synthesizers can be made, dependent upon the requirements of the individual system. However as an illustration a two loop system is shown in Fig. 6. This uses one loop to give the smaller steps and the second provides larger steps. This principle can be expanded to give wider ranges and smaller steps.

An example of a synthesizer using two loops

The first phase locked loop, PLL, has a digital divider and operates over the range 19 to 28 MHz. Having a reference frequency of 1 MHz it provides steps of 1 MHz. The signal from this loop is fed into the mixer of the second one. The second loop has division ratios of 10 to 19, but as the reference frequency has been divided by 10 to 100 kHz to give smaller steps.

The operation of the whole loop can be examined by looking at extremes of the frequency range. With the first loop set to its lowest value the divider is set to 19 and the output from the loop is at 19 MHz. This feeds into the second loop. Again this is set to the minimum value and the frequency after the mixer must be at 1.0 MHz. With the input from the first loop at 19 MHz this means that the VCO must operate at 20 MHz if the loop is to remain in lock.

At the other end of the range the divider of the first loop is set to 28, giving a frequency of 28 MHz. The second phase locked loop, PLL, has the divider set to 19, giving a frequency of 1.9 MHz between the mixer and divider. In turn this means that the frequency of the VCO must operate at 29.9 MHz. As the phase locked loops, PLLs, can be stepped independently it means that the whole synthesizer can move in steps of 100 kHz between the two extremes of frequency. As mentioned before this principle can be extended to give greater ranges and smaller steps, providing for the needs of modern receivers.

 

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Phase noise and frequency synthesizersOne of the main problems with frequency synthesisers and frequency synthesis

using phase locked loops is the fact that some designs generate high levels of phase noise. However it is possible to design some very good low phase noise synthesizers. The problem is often that receivers and transceivers are designed for low production costs, and this naturally means that some short cuts are needed.

What is phase noise?Phase noise is present on all signals to some degree and it is caused by small phase (and hence frequency) perturbations or jitter on the signal. It manifests itself as noise spreading out either side from the main carrier

Phase noise on a signal

Some signal sources are better than others. Crystal oscillators are very good and have very low levels of phase noise. Free running variable frequency oscillators normally perform well. Unfortunately synthesizers, and especially those based around phase locked loops, do not always fare so well unless they are well designed. If significant levels of phase noise are present on a synthesizer used as a local oscillator in a receiver, it can adversely affect the performance of the radio in terms of reciprocal mixing.

Some oscillators have phase noise levels that are quoted in their specifications. Any high quality signal generator will have the level of phase noise specified, as do many high performance crystal oscillators used as standards. Their performance is generally specified in dBc/Hz and at a given offset from the carrier. The term dBc simply refers to the level of noise relative to the carrier, i.e. -10 dBc means that the level is 10 lower than the carrier.

The bandwidth in which the noise is measured also has to be specified. The reason for this is that noise spreads over the frequency spectrum. Obviously the wider bandwidth that is used, the greater the level of noise that will pass through the filter and be measured. To prove this, just try selecting a different bandwidth on a receiver and check what happens to the noise level. It will rise for a wider bandwidth and fall when a narrow bandwidth is used. Technically the most convenient bandwidth to use a 1 Hz bandwidth and so this is used. When measuring this a wider bandwidth is usually used because it is difficult to obtain 1 Hz bandwidth filters and a correction is made mathematically.

Finally the level of noise varies as different offsets from the carrier are taken. Accordingly this must be included in a specification. A very good oscillator might have a specification of -100 dBc/Hz at 10 kHz offset.

It has already been mentioned that the level of phase noise changes as the offset from the carrier changes and for "simple" signal sources like crystal oscillators or variable frequency oscillators the phase noise reduces as the frequency from the main carrier is increased. For frequency synthesizers the picture is a little more complicated as we shall see.

Phase noise in synthesizersEach of the components in a frequency synthesizer produces noise that will contribute to the overall noise that appears at the output. The actual way in which the noise from any one element in the loop contributes to the output will depend upon where it is produced. Noise generated by the VCO will affect the output in a different way to that generated in the phase detector for example.

To see how this happens take the example of noise generated by the voltage controlled oscillator. This will pass through the divider chain and appear at the output of the phase detector. It will then have to pass through the loop filter. This will only allow through those components of the noise that are below the loop cut-off frequency. These will appear on the error voltage and have the effect of cancelling out the noise on the voltage controlled oscillator. As this effect will only take place within the loop bandwidth, it will reduce the level of noise within the loop bandwidth and have no effect on noise outside the loop bandwidth.

Noise generated by the phase detector is affected in a different way. Again only the components of the noise below the loop bandwidth will pass through the low pass filter. This means that there will be no components outside the loop bandwidth appearing on the tune voltage at the control terminal of the voltage controlled oscillator, and there will be no effect on the oscillator. Those components inside the loop bandwidth will appear at the oscillator control terminal. These will affect the oscillator and appear as phase noise on the output of the voltage controlled oscillator.

Matters are made worse by the fact that the division ratio has the effect of multiplying the noise level. This arises because the synthesizer effectively has the effect of multiplying the frequency of the reference. Consequently the noise level is also multiplied by a factor of 20 log N, where N is the division ratio.

Noise generated by the reference undergoes exactly the same treatments as that generated by the phase detector. It too is multiplied by the division ratio of the loop in the same way that the phase detector noise is. This means that even though the reference oscillator may have a very good phase noise performance this can be degraded significantly, especially if division ratios are high.

Dividers normally do not produce a significant noise contribution. Any noise they produce may be combined with that of the phase detector.

The combined noise of the loop at the output generally looks like that shown in Figure 2. Here it can be seen that the noise within the loop bandwidth arises from the phase detector and the reference. Outside the loop bandwidth it arises primarily from the voltage controlled oscillator. From this it can be seen that optimisation of the noise profile is heavily dependent upon the choice of the loop bandwidth. It is also necessary to

keep the division ratio in any loop down to reasonable levels. For example a 150 MHz synthesizer with a 12.5 kHz step size will require a division ratio of 12000. In turn this will degrade the phase detector and reference phase noise figures by 81 dB inside the loop bandwidth - a significant degradation by anyone's standards! Provided that division ratios are not too high then a wide loop bandwidth can help keep the voltage controlled oscillator noise levels down as well.

Noise profile of a typical synthesizer

Effects of phase noisePhase noise can have a number of effects. For SSB transmitters like those used for HF communications for ship to shore, amateur radio and other applications the main effect is that splatter appears either side of the main signal. This results from the phase noise either side of the signal will rising and falling in line with the amplitude variations of the main signal. For digital transmissions using frequency or phase modulation, the noise can introduce errors causing the bit error rate (BER) to rise.

For receivers the main problem is an effect known as reciprocal mixing. To look at how this occurs take the case of a superhet receiver tuned to a strong signal. The signal will pass through the radio frequency stages, and then in the mixer it will be mixed with the local oscillator to produce a new signal at the right frequency to pass through the IF filters. When the local oscillator is tuned away by ten kilohertz, for example the signal will no longer be able to pass through the IF filters. However it will still be possible for the phase noise on the local oscillator to mix with the strong incoming signal to produce a signal that will fall inside the receiver pass-band as shown in Figure 3. This could be sufficiently strong to mask out a weak station.

The way in which phase noise on a signal results in reciprocal mixing

SpecificationsA number of different methods are used to define the level of reciprocal mixing. Generally they involve the response of the receiver to a large off channel signal. To perform a reciprocal mixing measurement is rarely easy. The signal generator must always be much better than the receiver, otherwise the performance of the signal generator will be measured! To overcome this many people use an old valve generator because their performance is often very good in this respect.

A measurement can be made by noting the level of audio with a BFO on from a small signal. The signal is then tuned off channel by a given amount, normally about 20 kHz and then increased until the audio level rises to the same level as a result of the phase noise from the receiver. As the noise level is dependent upon the bandwidth of the receiver this has to be specified as well. Generally a bandwidth useable for SSB is used i.e. 2.7 kHz.

For example a good HF communications receiver might have a figure of 95 dB at a 20 kHz offset using a 2.7. kHz bandwidth. This figure will improve as the frequency offset from the main channel is increased. At 100 kHz one might expect to see a figure in excess of 105 dB or possibly more.

Another way of measuring the phase noise response is to inject a large signal into the receiver and monitor the level needed to give a 3 dB increase in background noise level.

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Direct digital synthesis (DDS)Direct digital synthesis (DDS) is a powerful technique used in the generation of

radio frequency signals for use in a variety of applications from radio receivers to signals generators and many more. The technique has become far more widespread in recent years with the advances being made in integrated circuit technology that allow much faster speeds to be handled which in turn enable higher frequency DDS chips to be made.

Although often used on its own, Direct Digital Synthesis is often used in conjunction with indirect or phase locked loop synthesizer loops. By combining both technologies it is possible to take advantage of the best aspects of each. In view of the fact that integrated circuits are now widely available, this makes them easy to use.

How it worksAs 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.

Block Diagram of a Basic Direct Digital Synthesizer (DDS).

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.

Operation of the phase accumulator in a direct digital synthesizer.

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.

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Graphical method for designing a PLL frequency synthesizer to meet a phase noise specification

- a simple graphical and understandable approach to understanding where phase noise is generated within a PLL frequency synthesizer and designing it to meet a requirement

Phase noise in PLL frequency synthesizers if of great importance because it determines many factors about the equipment into which it is incorporated. For receivers it determines the reciprocal mixing performance, and in some circumstances the bit error rate. In transmitters the phase noise performance of the frequency synthesizer determines features such as adjacent channel noise and it contributes to the bit error rate for the whole system.

Phase noise in a synthesizer loopPhase noise is generated at different points around the synthesizer loop and depending upon where it is generated it affects the output in different ways. For example, noise generated by the VCO has a different effect to that generated by the phase detector. This illustrates that it is necessary to look at the noise performance of each circuit block in the loop when designing the synthesizer so that the best noise performance is obtained.

Apart from ensuring that the noise from each part of the circuit is reduced to an absolute minimum, it is the loop filter which has the most effect on the final performance of the circuit because it determines the break frequencies where noise from different parts of the circuit start to affect the output.

To see how this happens take the example of noise from the VCO. Noise from the oscillator is divided by the divider chain and appears at the phase detector. Here it appears as small perturbations in the phase of the signal and emerges at the output of the phase detector. When it comes to the loop filter only those frequencies which are below its cut-off point appear at the control terminal of the VCO to correct or eliminate the noise. From this it can be seen that VCO noise which is within the loop bandwidth is attenuated, but that which is outside the loop bandwidth is left unchanged.

The situation is slightly different for noise generated by the reference. This enters the phase detector and again passes through it to the loop filter where the components below the cut-off frequency are allowed through and appear on the control terminal of the VCO. Here they add noise to the output signal. So it can be seen that noise from the reference is added to the output signal within the loop bandwidth but it is attenuated outside this.

Similar arguments can be applied to all the other circuit blocks within the loop. In practice the only other block which normally has any major effect is the phase detector and its noise affects the loop in exactly the same way as noise from the reference. Also if multi-loop synthesizers are used then the same arguments can be used again.

Effects of multiplicationAs noise is generated at different points around the loop it is necessary to discover what effect this has on the output. As a result it is necessary to relate all the effects back to the VCO. Apart from the different elements in the loop affecting the noise at the output in different ways, the effect of the multiplication in the loop also has an effect.

The effect of multiplication is very important. It is found that the level of phase noise from some areas is increased in line with the multiplication factor (i.e. the ratio of the

final output frequency to the phase comparison frequency). In fact it is increased by a factor of 20 log10 N where N is the multiplication factor. The VCO is unaffected by this, but any noise from the reference and phase detector undergoes this amount of degradation. Even very good reference signals can be a major source of noise if the multiplication factor is high. For example a loop which has a divider set to 200 will multiply the noise of the reference and phase detector by 46 dB.

From this information it is possible to build up a picture of the performance of the synthesizer. Generally this will look like the outline shown in Fig. 6. From this it can be seen that the noise inside the loop bandwidth is due mainly to components like the phase detector and reference, whilst outside the loop the VCO generates the noise. A slight hump is generally seen at the point where the loop filter cuts off and the loop gain falls to unity.

By predicting the performance of the loop it is possible to optimise the performance or look at areas which can be addressed to improve the performance of the whole synthesizer before the loop is even built. In order to analyse the loop further it is necessary to look at each circuit block in turn.

Voltage controlled oscillatorThe noise performance of the oscillator is of particular importance. This is because the noise performance of the synthesizer outside the loop is totally governed by its performance. In addition to this its performance may influence decisions about other areas of the circuit.

The typical noise outline for a VCO is flat at large frequency offsets from the carrier. It is determined largely by factors such as the noise figure of the active device. The performance of this area of the oscillator operation can be optimised by ensuring the circuit is running under the optimum noise performance conditions. Another approach is to increase the power level of the circuit so that the signal to noise ratio improves.

Closer in the noise starts to rise, initially at a rate of 20 dB per decade. The point at which this starts to rise is determined mainly by the Q of the oscillator circuit. A high Q circuit will ensure a good noise performance. Unfortunately VCOs have an inherently low Q because of the Q of the tuning varactors normally employed. Performance can be improved by increasing the Q, but this often results in the coverage of the oscillator being reduced.

Still further in towards the carrier the noise level starts to rise even faster at a rate of 30 dB per decade. This results from flicker or 1/f noise. This can be improved by increasing the level of low frequency feedback in the oscillator circuit. In a standard bipolar circuit a small un-bypassed resistor in the emitter circuit can give significant improvements.

To be able to assess the performance of the whole loop it is necessary to assess the performance of the oscillator once it has been designed and optimised. Whilst there are a number of methods of achieving this the most successful is generally to place the oscillator into a loop having a narrow bandwidth and then measure its performance with a spectrum analyser. By holding the oscillator steady this can be achieved relatively easily. However the results are only valid outside the loop bandwidth. However a test loop is likely to have a much narrower bandwidth than the loop being designed the noise levels in the area of interest will be unaltered.

ReferenceThe noise performance of the reference follows the same outlines as those for the VCO, but the performance is naturally far better. The reason for this is that the Q of the crystal is many orders of magnitude higher than that of the tuned circuit in the VCO.

Typically it is possible to achieve figures of -110 dBc/Hz at 10 Hz from the carrier and 140 dBc/Hz at 1 kHz from a crystal oven. Figures of this order are quite satisfactory for most applications. If lower levels of reference noise are required these can be obtain, but at a cost. In instances where large multiplication factors are necessary a low noise reference may be the only option. However as a result of the cost they should be avoided

wherever possible. Plots of typical levels of phase noise are often available with crystal ovens giving an accurate guide to the level of phase noise generated by the reference.

Frequency dividerDivider noise appears within the loop bandwidth. Fortunately the levels of noise generated within the divider are normally quite low. If an analysis is required then it will be found that noise is generated at different points within the divider each of which will be subject to a different multiplication factor dependent upon where in the divider it is generated and the division ratio employed from that point.

Most divider chains use several dividers and if an approximate analysis is to be performed it may be more convenient to only consider the last device or devices in the chain as these will contribute most to the noise. However the noise is generally difficult to measure and will be seen with that generated by the phase detector.

Phase detectorLike the reference signal the phase detector performance is crucial in determining the noise performance within the loop bandwidth. There are a number of different types of phase detector. The two main categories are analogue and digital.

Mixers are used to give analogue phase detectors. If the output signal to noise ratio is to be as good as possible then it is necessary to ensure that the input signal levels are as high as possible within the operating limits of the mixer. Typically the signal input may be limited to around -10 dBM and the local oscillator input to +10 dBm. In some instances higher level mixers may be used with local oscillator levels of +17 dBm or higher. The mixer should also be chosen to have a low NTR (noise temperature ratio). As the output is DC coupled it is necessary to have a low output load resistance to prevent a backward bias developing. This could offset the operation of the mixer and reduce its noise performance.

It is possible to calculate the theoretical noise performance of the mixer under optimum conditions. An analogue mixer is likely to give a noise level of around -153 dBc/Hz.

There are a variety of digital phase detectors which can be used. In theory these give a better noise performance than the analogue counterpart. At best a simple OR gate type will give figures about 10 dB better than an analogue detector and an edge triggered type (e.g. a dual D type or similar) will give a performance of around 5 dB better than the analogue detector.

These figures are the theoretical optimum and should be treated as guide although they are sufficient for initial noise estimates. In practice other factors may mean that the figures are different. A variety of factors including power supply noise, circuit layout etc. can degrade the performance from the ideal. If very accurate measurements are required then results from the previous use of the circuit, or from a special test loop can provide the required results.

Loop filterThere are a variety of parameters within the area of the loop filter which affect the noise performance of the loop. The break points of the filter and the unity gain point of the loop determined by the filter govern the noise profile.

In terms of contributions to the noise in the loop the major source is likely to occur if an operational amplifier is used. If this is the case a low noise variety must be used otherwise the filter will give a large contribution to the loop phase noise profile. This noise is often viewed as combined with that from the phase detector, appearing to degrade its performance from the ideal.

Plotting PerformanceHaving investigated the noise components from each element in the loop, it is possible to construct a picture of how the whole loop will perform. Whilst this can performed mathematically, a simple graphical approach quickly reveals an estimate of the performance and shows which are the major elements which contribute to the noise. In

this way some re-design can be undertaken before the design is constructed, enabling the best option to be chosen on the drawing board. Naturally it is likely to need some optimisation once it has been built, but this method enables the design to be made as close as possible beforehand.

First it is necessary to obtain the loop response. This is dependent upon a variety of factors including the gain around the loop and the loop filter response. For stability the loop gain must fall at a rate of 20 dB per decade (6 dB per octave) at the unity gain point. Provided this criterion is met a wide variety of filters can be used. Often it is useful to have the loop response rise at a greater rate than this inside the loop bandwidth. By doing this the VCO noise can be attenuated further. Outside the loop bandwidth a greater fall off rate can aid suppress the unwanted reference sidebands further. From a knowledge of the loop filter chosen the break points can be calculated and with a knowledge of the loop gain the total loop response can be plotted.

With the response known the components from the individual blocks in the loop can be added as they will be affected by the loop and seen at the output.

First take the VCO. Outside the loop bandwidth its noise characteristic is unmodified. However once inside this point the action of the loop attenuates the noise, first at a rate of 20 dB per decade, and then at a rate of 40 dB per decade. The overall affect of this is to modify the response of the characteristic as shown in Fig. 10. It is seen that outside the loop bandwidth the noise profile is left unmodified. Far out the noise is flat, but further in the VCO noise rises at the rate of 20 dB per decade. Inside the loop bandwidth the VCO noise will be attenuated first at the rate of 20 dB per decade, which in this case gives a flat noise profile. Then as the loop gain increases at the filter break point, to 40 dB per decade this gives a fall in the VCO noise profile of -20 dB per decade. However further in the profile of the stand alone VCO noise rises to -30dB per decade. The action of the loop gives an overall fall of -10 dB per decade.

The effects of the other significant contributions can be calculated. The reference response can easily be deduced from the manufacturers figures. Once obtained these must have the effect of the loop multiplication factor added. Once this has been calculated the effect of the loop can be added. Inside the loop there is no effect on the noise characteristic, however outside this frequency it will attenuate the reference noise, first at a rate of 20 dB per decade and then after the filter break point at 40 dB per decade.

The other major contributor to the loop noise is the phase detector. The effect of this is treated in the same way as the reference, having the effect of the loop multiplication added and then being attenuated outside the loop bandwidth.

Once all the individual curves have been generated they can be combined onto a single plot to gain a full picture of the performance of the synthesizer. When doing this it should be remembered that it is necessary to produce the RMS sum of the components because the noise sources are not correlated.

Once this has been done then it is possible to optimise the performance by changing factors like the loop bandwidth, multiplication factor and possibly the loop topology to obtain the best performance and ensure that the required specifications are met. In most cases the loop bandwidth is chosen so that a relatively smooth transition is made between the noise contributions inside and outside the loop. This normally corresponds to lowest overall noise situation.

SummaryAlthough this approach may appear to be slightly "low tech" in today's highly computerised engineering environment it has the advantage that a visual plot of the predicted performance can be easily put together. In this way the problem areas can be quickly identified, and the noise performance of the whole synthesizer optimised before the final design is committed.

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Radio receiver amplitude modulation (AM) demodulation

- using a simple diode detector ( demodulator )

One of the advantages of amplitude modulation (AM) is that it is cheap and easy to build a demodulator circuit for a radio receiver. The simplicity AM radio receivers AM is one of the reasons why AM has remained in service for broadcasting for so long. One of the key factors of this is the simplicity of the receiver AM demodulator.

A number of methods can be used to demodulate AM, but the simplest is a diode detector. It operates by detecting the envelope of the incoming signal. It achieves this by simply rectifying the signal. Current is allowed to flow through the diode in only one direction, giving either the positive or negative half of the envelope at the output. If the detector is to be used only for detection it does not matter which half of the envelope is used, either will work equally well. Only when the detector is also used to supply the automatic gain control (AGC) circuitry will the polarity of the diode matter.

The AM detector or demodulator includes a capacitor at the output. Its purpose is to remove any radio frequency components of the signal at the output. The value is chosen so that it does not affect the audio base-band signal. There is also a leakage path to enable the capacitor to discharge, but this may be provided by the circuit into which the demodulator is connected.

A simple diode detector (demodulator) for AM signals

This type of detector or demodulator is called a linear envelope detector because the output is proportional to the input envelope. Unfortunately the diodes used can introduce appreciable levels of harmonic distortion unless modulation levels are kept low. As a result these detectors can never provide a signal suitable for high quality applications.

Additionally these detectors ( demodulators ) are susceptible to the effects of selective fading experienced on short wave broadcast transmissions. Here the ionospheric propagation may be such that certain small bands of the signal are removed. Under normal circumstances signals received via the ionosphere reach the receiver via a number of different paths. The overall signal is a combination of the signals received via each path and as a result they will combine with each other, sometimes constructively to increase the overall signal level and sometimes destructively to reduce it. It is found that when the path lengths are considerably different this combination process can mean that small portions of the signal are reduced in strength. An AM signal consists of a carrier with two sidebands.

Spectrum of an amplitude modulated (AM) signal

If the section of the signal that is removed falls in one of the sidebands, it will change the tone of the received signal. However if carrier is removed or even reduced in strength, the signal will appear to be over modulated, and severe distortion will result. This is a comparatively common occurrence on the short waves, and means that diode detectors are not suitable for high quality reception. Synchronous demodulation ( detection ) is far superior.

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Synchronous demodulation / detectionToday's radio receivers offer very high levels of performance and boast many

facilities. Many radio receivers incorporate memories, phase locked loops, direct digital synthesis, digital signal processing and much more. One facility that can be very useful on the short wave bands is synchronous detection or synchronous demodulation as this can give much improved performance for receiving amplitude modulation (AM) transmissions. Unfortunately little is written about this form of modulation, and often it is a matter of accepting that it must be better than any normal options because it is included as a feature in the receiver specification.

Synchronous detection is used for the detection or demodulation of amplitude modulation (AM). This form of modulation is still widely used for broadcasting on the long, medium and short wave bands despite the fact that there are more efficient forms of modulation that can be used today. The main reason for its use nowadays is that it is very well established, and there are many millions of AM receivers around the world today.

In any receiver a key element is the detector. Its purpose is to remove the modulation from the carrier to give the audio frequency representation of the signal. This can be amplified by the audio amplifier ready to be converted into audible sound by headphones or a loudspeaker. Many receivers still use what is termed an envelope detector using a semiconductor diode for demodulating AM. These detectors have a number of disadvantages. The main one is that they are not particularly linear and distortion levels may be high. Additionally their noise performance is not particularly good at low signal levels.

These detectors also do not perform very well when the signal undergoes selective fading as often occurs on the short wave bands. An AM signal contains two sidebands and the carrier. For the signal to be demodulated correctly the carrier should be present at the required level. It can be seen that the signal covers a definite bandwidth, and the effects of fading may result in the carrier and possibly one of the sidebands being reduced in level. If this occurs then the received signal appears to be over-modulated with the result that distortion occurs in the demodulation process.

The spectrum of an amplitude modulated signal

Diode envelope detectorIn virtually every receiver a simple diode envelope detector is used. These circuits have the advantage that they are very simple and give adequate performance in many applications.

The circuit of a typical detector is shown in Figure 2. Here the diode first rectifies the signal to leave only the positive or negative going side of the signal, and then a capacitor removes any of the remaining radio frequency components to leave the demodulated audio signal. Unfortunately diodes are not totally linear and this is the cause of the distortion.

An envelope detector for AM signals

What is synchronous demodulationSignals can be demodulated using a system known as synchronous detection or demodulation. This is far superior to diode or envelope detection, but requires more circuitry. Here a signal on exactly the same frequency as the carrier is mixed with the incoming signal as shown in Figure 2. This has the effect of converting the frequency of the signal directly down to audio frequencies where the sidebands appear as the required audio signals in the audio frequency band.

The crucial part of the synchronous detector is in the production a local oscillator signal on exactly the same frequency as the carrier. Although it is possible to receive an AM signal without the local oscillator frequency on exactly the same frequency as the carrier this is the same as using the BFO in a receiver to resolve the signal. If the BFO is not exactly on the same frequency as the carrier then the resultant audio is not very good.

Synchronous demodulation

Fortunately this is not too difficult to achieve and although there are a number of ways of achieving this the most commonly used method is to pass some of the signal into a high gain limiting amplifier. The gain of the amplifier is such that it limits, and thereby removing all the modulation. This leaves a signal consisting only of the carrier and this can be used as the local oscillator signal in the mixer as shown in Fig. 4. This is most convenient, cheapest and certainly the most elegant method of producing synchronous demodulation.

A synchronous detector using a high gain-limiting amplifier to extract the carrier

Advantages of synchronous detectionA synchronous detector is more expensive to make than an ordinary diode detector when discrete components are used, although with integrated circuits being found in many receivers today there is little or no noticeable cost associated with its use as the circuitry is often included as part of an overall receiver IC.

Synchronous detectors are used because they have several advantages over ordinary diode detectors. Firstly the level of distortion is less. This can be an advantage if a better level of quality is required but for many communications receivers this might not be a problem. Instead the main advantages lie in their ability to improve reception under adverse conditions, especially when selective fading occurs or when signal levels are low.

Under conditions when the carrier level is reduced by selective fading, the receiver is able to re-insert its own signal on the carrier frequency ensuring that the effects of selective fading are removed. As a result the effects of selective fading can be removed to greatly enhance reception.

The other advantage is an improved signal to noise ratio at low signal levels. As the demodulator is what is termed a coherent modulator it only sees the components of noise that are in phase with the local oscillator. Consequently the noise level is reduced and the signal to noise ratio is improved.

Unfortunately synchronous detectors are only used in a limited number of receivers because of their increased complexity. Where they are used a noticeable improvement in receiver performance is seen and when choosing a receiver that will be used for short wave broadcast reception it is worth considering whether a synchronous detector is one of the facilities that is required.

 Frequency modulation is widely used in radio communications and broadcasting, particularly on frequencies above 30 MHz. It offers many advantages, particularly in mobile radio applications where its resistance to fading and interference is a great advantage. It is also widely used for broadcasting on VHF frequencies where it is able to provide a medium for high quality audio transmissions.

In view of its widespread use receivers need to be able to demodulate these transmissions. There is a wide variety of different techniques and circuits that can be sued including the Foster-Seeley, and ratio detectors using discreet components, and where integrated circuits are used the phase locked loop and quadrature detectors are more widely used.

What is FM?As the name suggests frequency modulation uses changes in frequency to carry the sound or other information that is required to be placed onto the carrier. As shown in Figure 1 it can be seen that as the modulating or base band signal voltage varies, so the frequency of the signal changes in line with it. This type of modulation brings several advantages with it. The first is associated with interference reduction. Much interference appears in the form of amplitude variations and it is quite easy to make FM receivers insensitive to amplitude variations and accordingly this brings about a reduction in the levels of interference. In a similar way fading and other strength variations in the signal have little effect. This can be particularly useful for mobile applications where charges in location as the vehicle moves can bring about significant signal strength changes. A further advantage of FM is that the RF amplifiers in transmitters do not need to be linear. When using amplitude modulation or its derivatives, any amplifier after the modulator must be linear otherwise distortion is introduced. For FM more efficient class C amplifiers may be used as the level of the signal remains constant and only the frequency varies.

Frequency modulating a signal

Wide band and Narrow bandWhen a signal is frequency modulated, the carrier shifts in frequency in line with the modulation. This is called the deviation. In the same way that the modulation level can be varied for an amplitude modulated signal, the same is true for a frequency modulated one, although there is not a maximum or 100% modulation level as in the case of AM.

The level of modulation is governed by a number of factors. The bandwidth that is available is one. It is also found that signals with a large deviation are able to support higher quality transmissions although they naturally occupy a greater bandwidth. As a result of these conflicting requirements different levels of deviation are used according to the application that is used.

Those with low levels of deviation are called narrow band frequency modulation (NBFM) and typically levels of +/- 3 kHz or more are used dependent upon the bandwidth available. Generally NBFM is used for point to point communications. Much higher levels of deviation are used for broadcasting. This is called wide band FM (WBFM) and for broadcasting deviation of +/- 75 kHz is used.

Receiving FMIn order to be able to receive FM a receiver must be sensitive to the frequency variations of the incoming signals. As already mentioned these may be wide or narrow band. However the set is made insensitive to the amplitude variations. This is achieved by having a high gain IF amplifier. Here the signals are amplified to such a degree that the amplifier runs into limiting. In this way any amplitude variations are removed.

In order to be able to convert the frequency variations into voltage variations, the demodulator must be frequency dependent. The ideal response is a perfectly linear voltage to frequency characteristic. Here it can be seen that the centre frequency is in the middle of the response curve and this is where the un-modulated carrier would be located when the receiver is correctly tuned into the signal. In other words there would be no offset DC voltage present.

The ideal response is not achievable because all systems have a finite bandwidth and as a result a response curve known as an "S" curve is obtained. Outside the badwidth of the system, the response falls, as would be expected. It can be seen that the frequency variations of the signal are converted into voltage variations which can be amplified by an audio amplifier before being passed into headphones, a loudspeaker, or passed into other electronic circuitry for the appropriate processing.

Characteristic "S" curve of an FM demodulator

To enable the best detection to take place the signal should be centred about the middle of the curve. If it moves off too far then the characteristic becomes less linear and higher levels of distortion result. Often the linear region is designed to extend well beyond the bandwidth of a signal so that this does not occur. In this way the optimum linearity is achieved. Typically the bandwidth of a circuit for receiving VHF FM broadcasts may be about 1 MHz whereas the signal is only 200 kHz wide.

DemodulatorsThere are a number of circuits that can be used to demodulate FM. Each type has its own advantages and disadvantages, some being used when receivers used discrete components, and others now that ICs are widely used.

Slope detectionThe very simplest form of FM demodulation is known as slope detection or demodulation. It simply uses a tuned circuit that is tuned to a frequency slightly offset from the carrier of the signal. As the frequency of the signal varies up and down in frequency according to its modulation, so the signal moves up and down the slope of the tuned circuit. This causes the amplitude of the signal to vary in line with the frequency variations. In fact at this point the signal has both frequency and amplitude variations. The final stage in the process is to demodulate the amplitude modulation and this can be achieved using a simple diode circuit. One of the most obvious disadvantages of this simple approach is the fact that both amplitude and frequency variations in the incoming signal appear at the output. However the amplitude variations can be removed by placing a limiter before the detector. Additionally the circuit is not particularly efficient as it operates down the slope of the tuned circuit. It is also unlikely to be particularly linear, especially if it is operated close to the resonant point to minimise the signal loss.

Ratio and Foster-Seeley detectorsWhen circuits employing discrete components were more widely sued, the Ratio and Foster-Seeley detectors were widely used. Of these the ratio detector was the most popular as it offers a better level of amplitude modulation rejection of amplitude

modulation. This enables it to provide a greater level of noise immunity as most noise is amplitude noise, and it also enables the circuit to operate satisfactorily with lower levels of limiting in the preceding IF stages of the receiver.

The operation of the ratio detector centres around a frequency sensitive phase shift network with a transformer and the diodes that are effectively in series with one another. When a steady carrier is applied to the circuit the diodes act to produce a steady voltage across the resistors R1 and R2, and the capacitor C3 charges up as a result.

The transformer enables the circuit to detect changes in the frequency of the incoming signal. It has three windings. The primary and secondary act in the normal way to produce a signal at the output. The third winding is un-tuned and the coupling between the primary and the third winding is very tight, and this means that the phasing between signals in these two windings is the same.

The primary and secondary windings are tuned and lightly coupled. This means that there is a phase difference of 90 degrees between the signals in these windings at the centre frequency. If the signal moves away from the centre frequency the phase difference will change. In turn the phase difference between the secondary and third windings also varies. When this occurs the voltage will subtract from one side of the secondary and add to the other causing an imbalance across the resistors R1 and R2. As a result this causes a current to flow in the third winding and the modulation to appear at the output.

The capacitors C1 and C2 filter any remaining RF signal which may appear across the resistors. The capacitor C4 and R3 also act as filters ensuring no RF reaches the audio section of the receiver.

The ratio detector

The Foster Seeley detector has many similarities to the ratio detector. The circuit topology looks very similar, having a transformer and a pair of diodes, but there is no third winding and instead a choke is used.

The Foster-Seeley detector

Like the ratio detector, the Foster-Seeley circuit operates using a phase difference between signals. To obtain the different phased signals a connection is made to the primary side of the transformer using a capacitor, and this is taken to the centre tap of the transformer. This gives a signal that is 90 degrees out of phase.

When an un-modulated carrier is applied at the centre frequency, both diodes conduct, to produce equal and opposite voltages across their respective load resistors. These voltages cancel each one another out at the output so that no voltage is present. As the carrier moves off to one side of the centre frequency the balance condition is destroyed, and one diode conducts more than the other. This results in the voltage across one of the resistors being larger than the other, and a resulting voltage at the output corresponding to the modulation on the incoming signal.

The choke is required in the circuit to ensure that no RF signals appear at the output. The capacitors C1 and C2 provide a similar filtering function.

Both the ratio and Foster-Seeley detectors are expensive to manufacture. Wound components like coils are not easy to produce to the required specification and therefore they are comparatively costly. Accordingly these circuits are rarely used in modern equipment.

Quadrature detectorAnother form of FM detector or demodulator that can be these days is called the quadrature detector. It lends itself to use with integrated circuits and as a result it is in widespread use. It has the advantage over the ratio and Foster-Seeley detectors that it only requires a simple tuned circuit.

For the quadrature detector, the signal is split into two components. One passes through a network that provides a basic 90 degree phase shift, plus an element of phase shift dependent upon the deviation and into one port of a mixer. The other is passed straight into another port of the mixer. The output from the mixer is proportional to the phase difference between the two signals, i.e. it acts as a phase detector and produces a voltage output that is proportional to the phase difference and hence to the level of deviation on the signal.

The detector is able to operate with relatively low input levels, typically down to levels of around 100 microvolts and it is very easy to set up requiring only the phase shift network to be tuned to the centre frequency of the expected signal. It also provides good linearity enabling very low levels of distortion to be achieved.

Often the analogue multiplier is replaced by a logic AND gate. The input signal is hard limited to produce a variable frequency pulse waveform. The operation of the circuit is fundamentally the same, but it is known as a coincidence detector. Also the output of the

AND gate has an integrator to "average" the output waveform to provide the required audio output, otherwise it would consist of a series of square wave pulses.

Phase locked loop (PLL)Another popular form of FM demodulator comes in the form of a phase locked loop. Like the quadrature detector, phase locked loops do not need to use a coil, and therefore they make a very cost effective form of demodulator.

The way in which they operate is very simple. The loop consists of a phase detector into which the incoming signal is passed, along with the output from the voltage controlled oscillator (VCO) contained within the phase locked loop. The output from the phase detector is passed into a loop filter and then sued as the control voltage for the VCO.

Phase locked loop (PLL) FM demodulator

With no modulation applied and the carrier in the centre position of the pass-band the voltage on the tune line to the VCO is set to the mid position. However if the carrier deviates in frequency, the loop will try to keep the loop in lock. For this to happen the VCO frequency must follow the incoming signal, and for this to occur the tune line voltage must vary. Monitoring the tune line shows that the variations in voltage correspond to the modulation applied to the signal. By amplifying the variations in voltage on the tune line it is possible to generate the demodulated signal.

It is found that the linearity of this type of detector is governed by the voltage to frequency characteristic of the VCO. As it normally only swings over a small portion of its bandwidth, and the characteristic can be made relatively linear, the distortion levels from phase locked loop demodulators are normally very low.

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Radio receiver filter options- including LC filter, crystal filter, mechanical filter, ceramic filter, and roofing filter

There is a wide variety of different types of filter used within superhet radios. Some radios will simply use filters made up from the tuned transformers (LC filters based on capacitors and inductors) linking the different intermediate frequency stages within the radios or used with an IC in the radio. Other radio receivers may incorporate highly selective crystal filters, whereas others may use mechanical filters (like those used by the Collins Radio Company some years ago) or ceramic filters. Each radio will have its own requirements, and the choice of filter to meet its needs of performance and cost.

LC tuned circuitsThe simplest type of filter is an ordinary L-C tuned circuit. In many older radio using discrete semiconductors, or older radios using vacuum tubes they take the form of transformers to couple the individual stages in an IF amplifier chain. Often there are two or three stages with tuned circuits. Using them it is usually possible to achieve sufficient selectivity for a medium wave AM or VHF FM broadcast radio. However for a good quality communications receiver it is rarely possible to be able to achieve the required degree of selectivity using just L-C filters.

In more modern radios using integrated circuits a single tuned circuit could be used in conjunction with an integrated, as the concept of inter-stage coupling is not employed in the same manner. Typically a ceramic filter, rather than an LC circuit is more likely to be used.

If L-C filters were used in a radio using interstage transformers then it would be possible to increase the degree of selectivity by increasing the number of tuned circuits between each stage. This is not ideal for a number of reasons. In the first case it increases the difficulty of aligning the set. In addition to this each tuned circuit will introduce a certain amount of loss. Increasing the number of tuned circuits will increase the amount of gain required, sometimes necessitating a further stage of gain. A further disadvantage is that it is not easy to alter the degree of selectivity by switching in additional L-C filters. If this is to be achieved then it is often preferable to switch in a further type of filter such as a crystal filter.

Crystal FiltersCrystal filters provide the main selectivity in of most of today's high performance sets. They provided exceedingly high degrees of selectivity which are hard to equal in terms of performance and cost.

The crystals in the filters are made from a substance called quartz. This is basically a form of crystalline silicon. Originally natural deposits were used to manufacture the crystals required for the electronics industry. Now quartz crystals are grown synthetically under controlled conditions to produce very high quality material.

The crystals use the piezo-electric effect for their operation. This effect occurs in a number of substances and it converts a mechanical stress into a voltage and vice versa. Many electrical transducers use the effect converting electrical impulses or signals into mechnical vibrations and vice versa.

In quartz crystal resonators the piezo-electric effect is used in conjunction with the mechanical resonances which occur in the substance. The electrical signals passing into the crystal are converted into mechanical vibrations which interact with the resonances of the crystal. In this way the crystal uses the piezo-electric effect to enable the mechanical resonances to tune the electrical signals. These mechanical resonances have exceedingly high Q factors. Many crystals will exhibit values of several thousand. This is many orders of magnitude higher than ordinary tuned filters made from conventional inductors and capacitors where values of a hundred or so are considered high. Typically the Q of an LC tuned circuit may be reach values of a few hundred. For quartz crystals values of Q may exceed 100 000.

Further details about quartz, its properties and the ways in which crystals are manufactured and used can be found on the Electronic components section of this site - see side menu for the link.

The response of a single crystal is too narrow for many applications. Normally a filter is required to have a passband, possibly of a few hundred Hertz, or a few kilohertz, and outside this bandwidth, other signals should be totally rejected. While it is not possible to achieve the perfect filter very high degrees of selectivity can be achieved. By adding several crystals together it is possible to obtain the performance that is required. Often crystal filters are referred to as having a certain number of poles. This terminology comes from the filter analysis design process, but effectively there is one crystal in the filter for every pole.

A two pole filter (i.e. one with two crystals) is not normally adequate to meet many requirements. The shape factor which is the ratio between the bandwidth where the stopband attenuation starts and the bandwidth of the passband) can be greatly improved by adding further sections. Typically ultimate rejections of 70 dB and more are required in a receiver. As a rough guide a two pole filter will generally give a rejection of around 20 dB; a four pole filter, 50 dB; a six pole filter, 70 dB; and an eight pole one 90 dB.

Monolithic filtersWith more items being integrated onto single chips these days it is hardly surprising to find that a similar approach is being adopted for crystal filters. Instead of having several separate or discrete crystals in a filter, even if they are all contained in the same can, it is possible to put a complete filter onto a single quartz crystal, hence the name monolithic crystal filter.

In essence the filter is made up by placing two sets of electrodes at opposite sides of a single AT cut crystal. The coupling between the two electrodes acts in such a way that a highly selective filter is produced.

Monolithic filters have only been available since the 1970s. Even now a large number of filter manufacturers do not produce them, preferring to use the more traditional filters made from individual crystals.

While it had been known for a long while that a two pole filter could be made up on a single crystal, the idea was not developed because the way in which it worked was not understood. After much work, scientists at Bell Laboratories in the USA discovered its mode of operation. Very simply it consists of two acoustically coupled resonators.

A monolithic crystal filter consists of a crystal blank onto which two sets of electrodes or plates are placed at opposite ends of the blank. Each set consists of an electrode on either side of the blank. When the electrical signal is placed across one pair of electrodes, the piezo-electric effect converts this into mechanical vibrations. These travel across the crystal to the other electrodes where they are converted back into an electrical signal again. However if the acoustic signal is to travel across the crystal then its frequency must match the resonance of the crystal.

Often these filters are manufactured for operation below about 30 MHz, because above these frequencies the manufacturing costs tend to rise. However manufacturing techniques are improving all the time it is possible to use them above this. If this is required then the normal way of accomplishing this is to use an overtone mode. This considerably increases the maximum possible frequencies, although the performance is not usually quite as good.

Monolithic filters are used in many areas now. They offer better performance than their discrete counterparts and they can be made smaller - a feature which is becoming increasingly important in today's miniaturised electronics industry. The main drawback of these filters is that they require very specialised equipment for their manufacture.

Ceramic filtersQuartz is not the only substance to exhibit the piezo-electric effect combined with a sharp resonance. A number of ceramics are also used successfully to perform this function. Although filters made from these ceramics are not nearly as selective as their higher quality quartz relatives, they are cheaper and offer great improvements over their L-C counterparts.

Ceramic filters are made from a specialised family of ceramics, and the elements for filters are normally in the form of a small disc. They operate in exactly the same way as crystal filters, the signal being linked to the mechanical resonances by the piezo-electric effect. Generally ceramic filters have a much wider bandwidth and a poorer shape factor than their crystal counterparts. As a result they are rarely used in high performance communications receivers as the primary form of filtering, although their performance has improved dramatically in recent years and some examples of ceramic filters offering exceedingly good levels of performance are available. As a result they find widespread use in broadcast receivers for AM and VHF FM reception and some wireless applications.

Mechanical filtersWhen high performance filters are needed there is another type which can be considered. Although not nearly as popular as crystal filters these days, mechanical filters found widespread use a number of years ago. The Collins Radio Company (now Rockwell

Collins) was a famous manufacturer of these devices, introducing their first designs in 1952, these filters are still manufactured.

In essence their operation is very similar to that of a crystal, although the various functions are performed by individual components within the filter. At either end of the filter assembly there are transducers which convert the signals from their electrical form to mechanical vibrations, and back again at the other end. These vibrations are applied to a series of discs which are mechanically resonant at the required frequency. Each of these discs has a Q of which can be about 5000 or more, and they are arranged close to one another but not touching to form a long cylinder. A number of coupling rods are attached to run along the side of the assembly to transfer the vibrations from one section to the next. By altering the amount of coupling between the sections and the resonance of each disc, the response of the overall unit can be tailored to meet the exact requirements.

Operation of these mechanical filters is normally confined to frequencies between about 50 and 500 kHz. Below these frequencies the discs become too large, whilst at the top end of the range they are too small to manufacture and mount in the filters with any degree of reliability. Apart from the limited frequency range the other disadvantage is that the resonant frequency of these filters drifts with temperature. However one of their main advantages is that exceedingly narrow bandwidths can be achieved relatively easily, and the low levels of intermodulation distortion they introduce. Additionally the costs of these devices have been reduced over the years and the number of resonators that can be used can be between 2 and 12 dependent upon the requirements.

Roofing filtersIn many receivers the main filtering occurs only after there have been many stages of amplification. This means that a strong signal which is outside the pass-band of the main receiver filter can cause overloading especially in the early IF stages before the filter. This occurs because the AGC does not see the signal and reduce the gain of the earlier stages to take account of it, or the operator may not be aware of the signal and reduce the RF gain if a control is available.

To overcome this problem a wider bandwidth filter is placed early on in the IF stages to reduce the level of any strong off channel signals. The main filtering, however, is still provided late on in the receiver by the main full specification filter.

Roofing filters are often found in multi-conversion superhet receivers where the main filter is found after two or possibly three conversion stages. The roofing filter can be placed soon after the first mixer to reduce the effects of any strong off-channel signals.

SummaryThere is a good selection of filters that can be used in radio receivers. The actual type that is eventually decided upon a balance of performance, cost and other factors. For many applications where the highest levels of performance are not needed, ceramic filters provide the ideal solution being very cheap and easy to use while providing levels of performance that are quite adequate for many applications. For applications where only the highest levels of performance are required, crystal filters are the most common solution either as units made from discrete crystals or as monolithic filters. However mechanical filters could ebb considered for some applications. These days LC filters are not widely used because the cost of winding coils is high, and often ceramic filters are more convenient, cheaper, and offer a better level of performance.

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Quartz crystal band pass filters- including the single crystal filter, half lattice filter and ladder filter for use in radio receivers

Crystal filters are widely used in many applications including radio receivers. The very high level of Q makes them ideal for use as the primary band pass filter in a radio receiver. As a result of this there are a number of circuits that have been used to provide the required level of selectivity and performance over the years. These include the single crystal filter, the half lattice crystal filter and the ladder filter.

Single crystal filterThe simplest crystal filter employs a single crystal. This type of filter was developed in the 1930s and was used in early receivers dating from before the 1960s but is rarely used today. Although it employs the very high Q of the crystal, its response is asymmetric and it is too narrow for most applications, having a bandwidth of a hundred Hz or less.

In the circuit there is a variable capacitor that is used to compensate for the parasitic capacitance in the crystal. This capacitor was normally included as a front panel control.

Diagram of filter using a single quartz crystal

Half lattice crystal filterThis form of band pass filter provided a significant improvement in performance over the single. In this configuration the parasitic capacitances of each of the crystals cancel each other out and enable the circuit to operate satisfactorily. By adopting a slightly different frequency for the crystals, a wider bandwidth is obtained. However the slope response outside the required pass band falls away quickly, enabling high levels of out of band rejection to be obtained. Typically the parallel resonant frequency of one crystal is designed to be equal to the series resonant frequency of the other.

Despite the fact that the half lattice crystal filter can offer a much flatter in-band response there is still some ripple. This results from the fact that the two crystals have different resonant frequencies. The response has a small peak at either side of the centre frequency and a small dip in the middle. As a rough rule of thumb it is found that the 3 dB bandwidth of the filter is about 1.5 times the frequency difference between the two resonant frequencies. It is also found that for optimum performance the matching of the filter is very important. To achieve this, matching resistors are often placed on the input and output. If the filter is not properly matched then it is found that there will be more in-band ripple and the ultimate rejection may not be as good.

Diagram of half lattice crystal filter

A two pole filter (i.e. one with two crystals) is not normally adequate to meet many requirements. The shape factor can be greatly improved by adding further sections. Typically ultimate rejections of 70 dB and more are required in a receiver. As a rough guide a two pole filter will generally give a rejection of around 20 dB; a four pole filter, 50 dB; a six pole filter, 70 dB; and an eight pole one 90 dB.

Crystal ladder filterFor many years the half lattice filter was possibly the most popular format used for crystal filters. More recently the ladder topology has gained considerable acceptance. In this form of crystal pass band filter all the resonators have the same frequency, and the inter-resonator coupling is provided by the capacitors placed between the resonators with the other termination connected to earth.

Four pole ladder crystal filter

 Although crystal filters are widely used as the high performance filters in receivers, mechanical filters are another option. Mechanical filters have been used for many years and are able to provide excellent service at frequencies up to just under 1 MHz. These filters can offer advantages over crystal filters in some instances being small, very stable, and rugged. In fact they are not subject to deterioration due to continuous exposure to shock and vibration, a factor that can be particularly important in some applications. A further advantage is that they offer low levels of intermodulation distortion, a factor that is often overlooked in many receiver designs.

Principles of operationIn essence their operation is very similar to that of a crystal, although the various functions are performed by individual components within the filter. At either end of the filter assembly there are transducers which convert the signals from their electrical form to mechanical vibrations, and back again at the other end. These vibrations are applied to a series of mechanical resonators which are mechanically resonant at the required frequency. The resonators are mechanically coupled, typically with coupling wires to transfer the vibrations from one section to the next. By altering the amount of coupling between the sections and the natural frequency of each resonator, the response of the overall unit can be tailored to meet the exact requirements.

Types of mechanical filterThere are several types of mechanical filter. The choice of the type depends upon the frequency in use and the application.

The first type is known as the torsional filter. This type of mechanical filter uses rods that vibrate in torsion. Electrical energy is coupled in by means of a piezoelectric ceramic transducer into torsional motion. These filters are used for frequencies in the range from below 100 kHz to just under 1 MHz.

Seven-resonator torsional mechanical filterImage Courtesy Rockwell Collins

A second type of filter is known as the bar flexural mode mechanical filter. This is used for low frequency designs, typically having a centre frequency between 5 to 100 kHz and with bandwidths of .2 to 1.5 percent.

Bar flexural mode mechanical filterImage Courtesy Rockwell Collins

SummaryAlthough mechanical filters are not used as widely as crystal filters, they can nevertheless offer an excellent solution in some instances. In view of this they are the ideal solution for many applications where high performance filters are required at frequencies below about 700 kHz to 1MHz.

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SAW filters

- an overview or tutorial about SAW filters used as RF and IF filters.

Surface Acoustic Wave (SAW) technology is used in many areas of electronics to provide resonators for oscillators, filters and transformers. One of the major uses of these devices is as SAW filters which find widespread use in radio applications. These SAW

filters provide good performance filtering solutions while offering a cost effective solution.

SAW filters are widely used in cell phone applications for filtering. Here they provide considerable advantages in terms of cost and size, in an environment where these two aspects are of considerable importance. Additionally their importance in the cellular industry has meant that considerable amounts of research and development have been undertaken on SAW filter s in the last 20 years, and their performance has improved considerably in this period.

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DSP - Digital Signal Processing tutorial- an overview or introduction to the basics of Digital Signal Processing, DSP, and how it can be used in radio receiver technology to improve performance and flexibility

Today, Digital Signal Processing, DSP, is widely used in radio receivers as well as in many other applications from television, radio transmission, or in fact any applications where signals need to be processed. Today it is not only possible to purchase digital signal processor integrated circuits, but also DSP cards for use in computers. Using these DSP cards it is possible to develop software or just use a PC platform in which to run the DSP card.

DSP has many advantages over analogue processing. It is able to provide far better levels of signal processing than is possible with analogue hardware alone. It is able to perform mathematical operations that enable many of the spurious effects of the analogue components to be overcome. In addition to this, it is possible to easily update a digital signal processor by downloading new software. Once a basic DSP card has been developed, it is possible to use this hardware design to operate in several different environments, performing different functions, purely by downloading different software. It is also able to provide functions that would not be possible using analogue techniques. For example a complicated signal such as Orthogonal Frequency Division Multiplex (OFDM) which is being used for many transmissions today needs DSP to become viable.

Despite this DSP has limitations. It is not able to provide perfect filtering, demodulation and other functions. There are mathematical limitations. In addition to this the processing power of the DSP card may impose some processing limitations. It is also more expensive than many analogue solutions, and thus it may not be cost effective in some applications. Nevertheless it has many advantages to offer, and with the wide availability of cheap DSP hardware and cards, it often provides an attractive solution for many radio applications.

What is DSP?As the name suggests, digital signal processing is the processing of signals in a digital form. DSP is based upon the fact that it is possible to build up a representation of the signal in a digital form. This is done by sampling the voltage level at regular time intervals and converting the voltage level at that instant into a digital number proportional to the voltage. This process is performed by a circuit called an analogue to digital converter, A to D converter or ADC. In order that the ADC is presented with a steady voltage whilst it is taking its sample, a sample and hold circuit is used to sample the voltage just prior to the conversion. Once complete the sample and hold circuit is ready to update the voltage again ready for the next conversion. In this way a succession of samples is made.

 

Sampling a waveform for DSP

 

Once in a digital format the real DSP is able to be undertaken. The digital signal processor performs complicated mathematical routines upon the representation of the signal. However to use the signal it then usually needs to be converted back into an analogue form where it can be amplified and passed into a loudspeaker or headphones. The circuit that performs this function is not surprisingly called a digital to analogue converter, D to A converter or DAC.

 

Block diagram of a Digital Signal Processor, DSP)

 

The advantage of DSP, digital signal processing is that once the signals are converted into a digital format they can be manipulated mathematically. This gives the advantage that all the signals can be treated far more exactly, and this enables better filtering, demodulation and general manipulation of the signal. Unfortunately it does not mean that filters can be made with infinitely steep sides because there are mathematical limitations to what can be accomplished.

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FPGAs for DSP Hardware

- the advantages and disadvantages of using FPGAs rather than DSP processors in the DSP hardware.

When designing the hardware system for a DSP application it is necessary to carefully consider the approach that will be taken. One of the fundamental decisions involves whether to use a standard DSP processor, or whether to use an FPGA in the DSP hardware. Each has its own advantages and they need to be carefully balanced at the earliest stages of the design.

DSP processorA DSP processor is a specialised processor that is designed specifically for operating

complex mathematically orientated intensive calculations very swiftly. As processing needs to be undertaken almost in real time, the speed of the processor is one of the main limiting performance criteria for the performance of the system For example very steep filters need more processing than those that are not so steep, etc..

While DSP processors, despite their sophistication in terms of processing have limitations, they also have advantages. One of these is in their cost. They may still be expensive by some standards, but they are nevertheless cheaper than their counterparts, the FPGA.

FPGAs for DSPThe other approach that many adopt is to use an FPGA as the core of the DSP hardware. These devices can be programmed and there are many set cores that can be used to provide the routines that are required. For example if a filter is required, then it is possible to tailor circuitry within the FPGA to undertake this. Similarly other functions can be programmed in on top of the basic processor. In this way the FPGA is able to be programmed to provide a highly efficient and tailored solution.

The main disadvantage of the FPGA is its cost. FPGAs are more costly that DSP processors and therefore performance has to be weighed against cost.

SummaryFPGAs and DSP processors provide two very different approaches to the design of DSP hardware systems. Each have their own advantages. There are many high sampling rate applications that an FPGA does easily, while the DSP could not. Equally, there are many complex software problems that the FPGA cannot address.