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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. When 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.
This lists modes, with modulation and demodulation systems.
|AM||Plate, etc.||Envelope Detector|
|PM & FM||Reactance||Discriminator,|
Balanced with Filter
To general AM the tradition method was to mix a high level audio signal into the high-level RF signal, such as by placing a 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.
SSB can be generated using a balanced modulator, after which the unwanted sideband is removed, often using a crystal 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.
These are mathematical models of electronic signal processes. They can be used to digitally analyse or manipulate signals.
The Fourier transform allows things such ash 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, as well as the original In-phase signal, a 90 degree phase-shifted version of the signal is created, called the Quadrature. These are termed the I and Q.
These are the actual questions from the Extra licence exam pool, as published by the NCVEC.
Which of the following can be used to generate FM phone emissions?
A. A balanced modulator on the audio amplifier
B. A reactance modulator on the oscillator
C. A reactance modulator on the final amplifier
D. A balanced modulator on the oscillator
A reactance modulator, importantly located in the oscillator, generates FM, answer B.
What is the function of a reactance modulator?
A. To produce PM signals by using an electrically variable resistance
B. To produce AM signals by using an electrically variable inductance or capacitance
C. To produce AM signals by using an electrically variable resistance
D. To produce PM or FM signals by using an electrically variable inductance or 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.
What is a frequency discriminator stage in a FM receiver?
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 detecting FM signals, answer D.
What is one way a single-sideband phone signal can be generated?
A. By using a balanced modulator followed by a filter
B. By using a reactance modulator followed by a mixer
C. By using a loop modulator followed by a mixer
D. By driving 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.
What circuit is added to an FM transmitter to boost the higher audio frequencies?
A. A de-emphasis network
B. A heterodyne suppressor
C. An audio prescaler
D. A pre-emphasis network
This is the pre-emphasis network, 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.
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 frequency modulation of high frequency audio, so this compensates for that, answer A.
What is meant by the term "baseband" in radio communications?
A. The lowest frequency band that the transmitter or receiver covers
B. The frequency components present in the modulating signal
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, or perhaps to the frequency of its components. 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.
What are the principal frequencies that appear at the output of a mixer circuit?
A. Two and four times the original frequency
B. The sum, difference and square root of the 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.
What occurs when an excessive amount of signal energy reaches a mixer circuit?
A. Spurious mixer products are generated
B. Mixer blanking occurs
C. Automatic limiting occurs
D. A beat frequency is generated
Excessive often implies something bad is going to result, in this case, the generation of spurious mixer products, answer A.
How does a diode 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.
Which type of detector is used for demodulating SSB signals?
B. Phase detector
C. Product detector
D. Phase comparator
A product detector, answer C.
What is meant by direct digital conversion as applied to 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.
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.
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.
What is a common method of generating 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. Emulation of quartz crystal filter characteristics
D. Combine signals with a quadrature phase relationship
DSP can combine signals using a quadrature (90°) phase relationship, answer D.
How frequently must an analog signal be sampled by an analog-to-digital converter so that the signal can be accurately reproduced?
A. At half the rate of the highest frequency component of the signal
B. At 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.
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.
What function can a Fast Fourier Transform perform?
A. Converting analog signals to digital form
B. Converting digital signals to analog form
C. Converting digital signals from the time domain to the frequency domain
D. Converting 8-bit data to 16 bit data
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.
What is the function of decimation with regard to digital filters?
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 by discarding all but one sample in every 6. Answer B.
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.
What aspect of receiver analog-to-digital conversion determines the maximum receive bandwidth of a Direct Digital Conversion SDR?
A. Sample rate
B. Sample width in bits
C. Sample clock phase noise
D. Processor latency
This is the sample rate, answer A.
What sets the minimum detectable signal level for an SDR in the absence of atmospheric or thermal noise?
A. Sample clock phase 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.
Which of the following is an advantage of a Finite Impulse Response (FIR) filter vs an Infinite Impulse Response (IIR) digital filter?
A. FIR filters 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.
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.
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. Complex phasor representations
D. Double-precision math routines
More taps provide sharper filters, answer B.
On to: Practical Circuits 4 - Op-amps, Filters & Oscillators
You can find links to lots more on the Learning Material page.
Written by Julian Sortland, VK2YJS & AG6LE, June 2022.
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