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Frequency and phase modulation are two closely related forms of modulation, termed angle modulation.
Many older commercial and government radios which are nominally FM are in fact PM, as are some amateur devices. PM signals are created at a lower RF frequency, such as just over 12 MHz, then multiplied by 12; or just over 9 MHz, and multiplied by 16, we get to 2 metres, a little over 144 MHz. Some devices use "True FM", and may include this point in their marketing.
The demodulator, or means to recover the intelligence, is most often a discriminator, specifically a Foster-Seeley discriminator. In an FM audio receiver further audio processing may be used, including de-emphasis and other filters. AFSK signals such as AX-25 Packet including APRS can pass through this, but for 9600 bit per second packet a port which provides the output of the discriminator. This is normally on a 6-pin mini-DIN socket. See: Wikipedia: Foster Seeley Discriminator and Wikipedia: 9600_port.
This lists modes, with modulation and demodulation systems.
| Mode | Modulator | De-mododulator |
|---|---|---|
| AM | Plate, etc. | Envelope Detector |
| PM & FM | Reactance | Discriminator, PLL |
| SSB | Phase, Balanced with Filter | Product Detector |
To generate AM the tradition method was to mix a high level audio signal into the high-level RF signal, such as by placing a transformer between the positive supply rail and the plate of the final tube / valve. There are solid state versions of these, used in AM CBs. The alternative is to generate AM at a low level (including by DSP) and then amplify this, as in SSB transmitters.
A balanced modulator outputs the sum and difference between the two inputs is presented to the output, but not the input frequencies. If both input frequencies are suppressed the term "doubly balanced" is applied for clarity. Examples are the the NE602 / SA602A, or the newer NE612 / SA612A ICs, noting that these including an oscillator section. The "ring modulator", consisting of 4 signal diodes, also does this. Like the diode rectification bridge they can be drawn in a square, diamond, or X format. In countries where voice on 3.58 MHz is merely discouraged people build various DSB transmitters using an NTSC "Colorburst" crystal.
SSB can be generated using a balanced modulator, after which the unwanted sideband is removed, often using a crystal lattice filter. Phase based circuits, and DSP can also be used.
SSB is detected using a product detector, which can also be used for AM. Traditionally AM was detected using a diode, which converts the envelope of the AM signal back to an audio waveform.
Transforms are essentially mathematical models, including of electronic signal processes. They can be used to digitally analyse or manipulate signals.
The Fourier transform allows things such as analysing the frequencies which make up a musical note, or a square wave. The Fast Fourier Transform (FFT) is described as "Converting digital signals from the time domain to the frequency domain" in the exam. Feeding a square wave into a FFT system will display a fundamental and its odd harmonics, progressively reducing in level.
See: Wikipedia: Fourier transform and Wikipedia: Fast Fourier transform
Meanwhile, the Hilbert transform can be used to produce single-sideband signals in digital signal processor based systems. This has some heavy-duty maths regarding it: MathWorks: Single Sideband Modulation via the Hilbert Transform
For DSP filters to analyse the frequency of components of a signal, and decide what to do with them, they need to work with a large number of samples. The exam refers to these as taps. The benefit is sharper filters, but the cost is the need for more memory, and a longer delay.
The "DSP Guru" website includes many topics for those who wish to read further. This article covers Decimation and this one Finite Impulse Response.
Removed from this version of the exam, in many digital systems, the original "In-phase" signal is used in addition to a 90 degree phase-shifted version of the signal, called the Quadrature. These are termed the I and Q.
Decimation is a digital process which redised the sample rate, nominally just by dumping smaples, but in reality by filtering and trying to provide as close a representation as possible. This is done in the digital domain, but playing a CD into the hold music input of a 'phone system would do this in the analogue domain once it reaches the telephones network which runs at 8000 samples per second.
These are the actual questions from the Extra licence exam pool, as published by the NCVEC, for use until teh end of June 2028.
E7E01
Which of the following can be used to generate FM phone emissions?
A. A balanced modulator on the audio amplifier
B. Reactance modulation of a local oscillator
C. A reactance modulator on the final amplifier
D. A balanced modulator on the oscillator
A reactance modulator, importantly located in the (local) oscillator, generates FM, answer B.
E7E02
What is the function of a reactance modulator?
A. Produce PM or FM signals by varying a resistance
B. Produce AM signals by varying an inductance
C. Produce AM signals by varying a resistance
D. Produce PM or FM signals by varying a capacitance
Reactance modulation generates an angle modulation of some sort, which includes PM and FM. And reactance strongly implies an inductance or capacitance, put them together, and get answer D.
E7E03
What is a frequency discriminator?
A. An FM generator circuit
B. A circuit for filtering two closely adjacent signals
C. An automatic band-switching circuit
D. A circuit for detecting FM signals
These detect (demodulate) FM signals, answer D.
E7E04
What is one way to produce a single-sideband phone signal?
A. Use a balanced modulator followed by a filter
B. Use a reactance modulator followed by a mixer
C. Use a loop modulator followed by a mixer
D. Use a product detector with a DSB signal
A balanced modulator generates DSB, without a carrier. Adding a filter removes one of the sidebands, generating SSB, answer A.
E7E05
What is added to an FM speech channel to boost the higher audio frequencies?
A. A de-emphasis network
B. A harmonic enhancer
C. A heterodyne enhancer
D. A pre-emphasis network
A pre-emphasis network is added into the speech processing chain in the transmitter, answer D.
By doing this the high tones, where voice levels may be lower are boosted, to better compete with noise. The the far end de-emphasis circuit reduces the high-end audio to the correct level, along with any noise in that frequency range. This was pretty much what Dolby settings on cassettes did.
E7E06
Why is de-emphasis commonly used in FM communications receivers?
A. For compatibility with transmitters using phase modulation
B. To reduce impulse noise reception
C. For higher efficiency
D. To remove third-order distortion products
It appears that phase modulation results in a greater level of modulation of high frequency audio, so this compensates for that, answer A.
This may be a non-ideal answer, but is what gets you the banana.
E7E07
What is meant by the term "baseband" in radio communications?
A. The lowest frequency band that the transmitter or receiver covers
B. The frequency range occupied by a message signal prior to modulation
C. The unmodulated bandwidth of the transmitted signal
D. The basic oscillator frequency in an FM transmitter that is multiplied to increase the deviation and carrier frequency
This refers to the modulating signal. An example is "baseband video", also termed composite, which is is what comes out of a yellow RCA socket. If it is NTSC, then it has various components up to around 6 MHz, a little more for PAL. This differs from the modulated signal which comes from a F or Belling Lee socket on a VCR / VHS or set top box, to go the the TV's antenna input, which is somewhere between around 40 and 800 MHz, depending on the channel used. Answer B.
E7E08
What are the principal frequencies that appear at the output of a mixer?
A. Two and four times the input frequency
B. The square root of the product of input frequencies
C. The two input frequencies along with their sum and difference frequencies
D. 1.414 and 0.707 times the input frequency
If we modulate a 1 MHz signal with a 1 kHz tone, we get 1 kHz, 1 MHz, plus 999 kHz and 1.001 MHz, which you can see are the original frequencies, and the sum and difference of these, answer C.
E7E09
What occurs when the input signal levels to a mixer are too high?
A. Spurious mixer products are generated
B. Mixer blanking occurs
C. Automatic limiting occurs
D. Excessive AGC voltage levels are generated
This causes the generation of spurious mixer products, answer A.
E7E10
How does a diode envelope detector function?
A. By rectification and filtering of RF signals
B. By breakdown of the Zener voltage
C. By mixing signals with noise in the transition region of the diode
D. By sensing the change of reactance in the diode with respect to frequency
A detector diode rectifies the RF signal to recover content on an AM signal, answer A.
E7E11
Which type of detector is used for demodulating SSB signals?
A. Discriminator
B. Phase detector
C. Product detector
D. Phase comparator
A product detector, answer C.
E7F01
What is meant by "direct sampling" in software defined radios?
A. Software is converted from source code to object code during operation of the receiver
B. Incoming RF is converted to a control voltage for a voltage controlled oscillator
C. Incoming RF is digitized by an analog-to-digital converter without being mixed with a local oscillator signal
D. A switching mixer is used to generate I and Q signals directly from the RF input
Typically for MF or low HF signals, a device such as a composite video A to D device for a PC can be used to digitise a section of thr RF spectrum directly. Dedicated SDRs have a greater sample rate, and can digitise the whole HF spectrum directly, for example, answer C.
A mixer and local oscillator are not required.
E7F02
What kind of digital signal processing audio filter is used to remove unwanted noise from a received SSB signal?
A. An adaptive filter
B. A crystal-lattice filter
C. A Hilbert-transform filter
D. A phase-inverting filter
This is the adaptive filter, which reacts automatically to interference such as noise and hetrodynes, answer A.
E7F03
What type of digital signal processing filter is used to generate an SSB signal?
A. An adaptive filter
B. A notch filter
C. A Hilbert-transform filter
D. An elliptical filter
This is the Hilbert-transform filter, answer C.
E7F04
Which method generates an SSB signal using digital signal processing?
A. Mixing products are converted to voltages and subtracted by adder circuits
B. A frequency synthesizer removes the unwanted sidebands
C. Varying quartz crystal characteristics are emulated in digital form
D. Signals are combined in quadrature phase relationship
DSP can combine signals using a quadrature (90°) phase relationship, answer D.
E7F05
How frequently must an analog signal be sampled to be accurately reproduced?
A. At least half the rate of the highest frequency component of the signal
B. At least twice the rate of the highest frequency component of the signal
C. At the same rate as the highest frequency component of the signal
D. At four times the rate of the highest frequency component of the signal
The sampling must be at twice the highest frequency, answer B.
E7F06
What is the minimum number of bits required for an analog-to-digital converter to sample a signal with a range of 1 volt at a resolution of 1 millivolt?
A. 4 bits
B. 6 bits
C. 8 bits
D. 10 bits
We need 1000 levels to do this, and the nearest binary value above this is 210 = 1024, so 10 bits. We could set the reference at 1.024 volts, so each step is 1 mV; or at 1 volt, and the steps would be 1/1024 volts, or 976.5625 μV. Answer D.
E7F07
What function is performed by a Fast Fourier Transform
A. Converting analog signals to digital form
B. Converting digital signals to analog form
C. Converting signals from the time domain to the frequency domain
D. Converting signals from the frequency domain to the time domain
FFT is a used to examine the frequency components of a waveform, answer C.
A modular oscilloscope may have an FFT module added to perform frequency analysis, and it can be a function is a DSO.
E7F08
What is the function of decimation?
A. Converting data to binary code decimal form
B. Reducing the effective sample rate by removing samples
C. Attenuating the signal
D. Removing unnecessary significant digits
Suppose we sample audio at 48 kHz using a USB microphone, and wish to send the audio to a 8 kHz system. Part of the process is digitally filtering the audio signal down to 4 kHz, then reducing the sample rate to 8000 samples per second, essentially by discarding all but one sample in every 6. Answer B.
E7F09
Why is an anti-aliasing digital filter required in a digital decimator?
A. It removes high-frequency signal components which would otherwise be reproduced as lower frequency components
B. It peaks the response of the decimator, improving bandwidth
C. It removes low-frequency signal components to eliminate the need for DC restoration
D. It notches out the sampling frequency to avoid sampling errors
This removes high frequency components which would appear as beat signals at lower frequencies, answer A.
E7F10
What aspect of receiver analog-to-digital conversion determines the maximum receive bandwidth of a direct-sampling software defined radio (SDR)?
A. Sample rate
B. Sample width in bits
C. Integral non-linearity
D. Differential non-linearity
This is the sample rate, answer A.
E7F11
What sets the minimum detectable signal level for a direct-sampling software defined receiver in the absence of atmospheric or thermal noise?
B. Reference voltage level and sample width in bits
C. Data storage transfer rate
D. Missing codes and jitter
The smallest signal is a function of the value of a single bit, and if we have n bits, the value of a bit is: VREF / 2 n. So, if the reference is 2mV and we have 12 bits: 0.002 / 212 = .002 / 4096 = 0.488 μV. Answer B.
E7F12
Which of the following is generally true of Finite Impulse Response (FIR) filters?
A. FIR filters can delay all frequency components of the signal by the same amount
B. FIR filters are easier to implement for a given set of passband rolloff requirements
C. FIR filters can respond faster to impulses
D. All of these choices are correct
FIR filters delay all frequency components of the signal by the same amount, answer A.
E7F13
What is the function of taps in a digital signal processing filter?
A. To reduce excess signal pressure levels
B. Provide access for debugging software
C. Select the point at which baseband signals are generated
D. Provide incremental signal delays for filter algorithms
These provide an "image" of the signal at multiple points over time, necessary for the filter algorithms, answer D.
E7F14
Which of the following would allow a digital signal processing filter to create a sharper filter response?
A. Higher data rate
B. More taps
C. Lower Q
D. Double-precision math routines
More taps provide sharper filters, answer B.
In the the early 1980s Audiovox and Tandy / RadioShack sold a manually tuned FM converter which tuned across the VHF-FM band, and downshifted the selected station to a particular frequency in the middle of the AM broadcast band. Reportedly these relied on slope detection of FM signal using the AM detector of the existing radio.
The term baseband also applies to certain digital data transmission systems using telephone company pairs, an example being Telecom Australia's DDN (digital data network), where the Network Terminal Units imposed digital signals onto the line, with data rates of 2400, 4800, or 9600 bps. These were NOT modems, and they did NOT modulate the signals at audio frequencies. They might be used in an insurance company branch, or perhaps at a branch library where text based screens (white, green, or orange on black) could display catalogue information at an acceptable rate, given the typical record is a few hundred bytes. Of course, this gave a slower-that-dialup experience when they went to a PC which displaying the catalogue in web page form. ISDN uses a similar method, with greater speed.
On to: Practical Circuits 4 - Op-amps, Filters & Oscillators
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Written by Julian Sortland, VK2YJS & AG6LE, April 2026.
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