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G7C

PRACTICAL CIRCUITS

- Transceiver design; filters; oscillators; digital signal processing (DSP)

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G7C011 of 14

What circuit is used to select one of the sidebands from a balanced modulator?

Why A balanced modulator suppresses the carrier but leaves both sidebands (double sideband suppressed carrier). To get single sideband you must remove one of them, and that is done with a sharp bandpass filter, typically a crystal or mechanical filter a few kHz wide placed at the IF. Shifting the carrier oscillator to the other side of the filter passband is how the rig swaps between USB and LSB.
Watch out The carrier oscillator feeds the balanced modulator and sets which sideband lands in the filter passband, but it does not itself do the selecting; IF and RF amplifiers only add gain, not selectivity of that sharpness.
Balanced modulator kills the carrier, the filter kills the extra sideband.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C022 of 14

What output is produced by a balanced modulator?

Why A balanced modulator is a mixer whose circuit balance cancels the carrier, so only the sum and difference products of the audio and the carrier oscillator appear at the output. That gives double-sideband suppressed-carrier RF, the first stage in generating SSB. A filter or phasing network then removes one sideband to leave single sideband.
Watch out The audio answers describe detectors or audio processors, not a modulator; a balanced modulator takes audio in and puts RF out. Frequency modulation is produced by a reactance modulator or a phase modulator, not by a balanced mixer.
Balanced means the carrier is balanced out: what's left is both sidebands, DSB suppressed carrier.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C033 of 14

What is one reason to use an impedance matching transformer at a transmitter output?

Why Modern solid state transmitters are designed to deliver full rated power into a 50 ohm resistive load, and they reduce output (or shut down) when the SWR rises. An impedance matching transformer or antenna tuner transforms whatever impedance appears at the end of the feed line into the 50 ohms the transmitter wants to see, allowing maximum power transfer. Note that it matches at the transmitter end; it does not change the SWR out on the feed line between the tuner and the antenna.
Watch out Reducing power output is what happens when the match is bad, not the purpose of the matching device, and radiation resistance is a property of the antenna itself that no transformer at the transmitter can change. Power supply ripple is a filtering issue in the DC supply, unrelated to RF impedance.
Transmitters want to see 50 ohms; the matching transformer is there to give them exactly that.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C044 of 14

How is a product detector used?

Why A product detector is just a mixer: it multiplies the incoming IF signal by a locally generated carrier from a beat frequency oscillator (BFO), and the difference frequency that falls out is the recovered audio. SSB and CW signals have no carrier of their own, so the BFO supplies the missing carrier that the detector needs to demodulate them. A simple envelope (diode) detector cannot do this, which is why SSB and CW receivers use product detectors.
Watch out The choice about detecting spurious mixing products is a play on the word 'product' in the name; the 'product' here refers to the mathematical multiplication of two signals, not to unwanted mixer outputs. FM receivers use discriminators or ratio detectors, not product detectors.
Product = multiply: IF times BFO equals audio. It is the SSB/CW demodulator.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C055 of 14

Which of the following is characteristic of a direct digital synthesizer (DDS)?

Why A direct digital synthesizer builds a waveform numerically: a phase accumulator steps through a sine lookup table and feeds a DAC, all clocked by a single crystal reference. Because the output frequency is just a fixed ratio of that clock, set by the tuning word, the accuracy and drift of the output are tied directly to the crystal. That gives you a continuously tunable VFO whose stability matches a crystal oscillator, which is exactly what a modern transceiver needs.
Watch out The choice about a pure sine wave is tempting, but a DDS output is a stair-stepped approximation containing DAC quantization noise and spurious products, so it needs low-pass filtering to clean it up.
DDS = crystal clock in, any frequency out. Crystal stability with VFO flexibility, but spurs need filtering.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C066 of 14

Which of the following is an advantage of a digital signal processing (DSP) filter compared to an analog filter?

Why A DSP filter is defined by software coefficients rather than by physical inductors, capacitors or crystals, so the same hardware can produce almost any bandwidth, center frequency and response shape just by loading different numbers. That lets a radio offer continuously variable passband width, steep brick-wall skirts, notch filters and noise reduction from one converter and processor. An analog filter's response is fixed by the parts soldered in, so each bandwidth needs its own filter.
Watch out The claim about reducing mixing products confuses filtering with mixer design; mixing products come from the nonlinear mixing stage, and DSP filtering usually happens after conversion to a low IF or baseband, not at VHF as the frequency choice suggests.
DSP filters are made of math, not parts: change the numbers, change the bandwidth and shape.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C077 of 14

What term specifies a filter's attenuation inside its passband?

Why Every real filter has some loss even at the frequencies it is supposed to pass, caused by resistive losses in its inductors, capacitors or crystals. That passband loss is called insertion loss because it is measured as the drop in signal level when the filter is inserted into the line compared with a straight-through connection. It is quoted in dB, and a good HF crystal or LC filter might show only 1 to 3 dB.
Watch out Ultimate rejection is the tempting one, but it describes attenuation far outside the passband, in the stopband, not inside it. Return loss describes how much power is reflected back due to an impedance mismatch, and Q describes the sharpness or selectivity of the response.
Insert the filter, lose some signal: insertion loss is inside the passband, ultimate rejection is outside it.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C088 of 14

Which parameter affects receiver sensitivity?

Why Sensitivity is the smallest signal a receiver can usefully detect, and that depends on the whole signal-to-noise picture, not one part. Front-end gain sets how far the incoming signal is raised above the noise contributed by later stages, the noise figure of that first amplifier sets the noise floor the signal has to beat (noise added early gets amplified along with the signal), and the bandwidth at the demodulator determines how much noise power reaches the detector since noise power is proportional to bandwidth. Tighten the bandwidth, lower the noise figure, or add appropriate low-noise gain up front and the minimum discernible signal improves.
Watch out Noise figure alone is the most tempting single pick because it is the classic sensitivity spec, but gain and detection bandwidth change the measured MDS just as directly.
Sensitivity = signal vs noise: gain lifts the signal, noise figure adds noise, bandwidth lets noise in. All three count.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C099 of 14

What is the phase difference between the I and Q RF signals that software-defined radio (SDR) equipment uses for modulation and demodulation?

Why I and Q stand for In-phase and Quadrature, and "quadrature" means a quarter of a cycle, or 90 degrees. In an SDR the incoming signal is mixed with two copies of the local oscillator that are 90 degrees apart, producing two baseband streams whose relative phase lets the software determine both amplitude and phase (and therefore which side of the carrier a signal falls on). That extra phase information is what allows one pair of mixers to handle SSB, CW, FM and digital modes in software.
Watch out 180 degrees is the phase relationship in a push-pull or balanced circuit, where the two signals are simply inverted; that would give no new information about the signal, since a 180 degree copy is just the negative of the original.
Q = quadrature = a quarter of a cycle = 90 degrees.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C1010 of 14

What is an advantage of using I-Q modulation with software-defined radios (SDRs)?

Why I-Q modulation splits a signal into two components 90 degrees apart, in-phase (I) and quadrature (Q). Any signal can be described by its amplitude and phase at each instant, and any amplitude/phase pair can be built from the right combination of I and Q values. Because software can compute those I and Q streams however it likes, an SDR can generate or decode AM, FM, SSB, CW, PSK, or any other mode with nothing but a change in the processing code.
Watch out The choice about eliminating high resolution A/D converters has it backwards: good dynamic range in an SDR still depends heavily on the resolution of the converters, and I-Q sampling does not change that.
I and Q give you amplitude and phase, and amplitude plus phase equals every mode.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C1111 of 14

Which of these functions is performed by software in a software-defined radio (SDR)?

Why In a software-defined radio the signal is digitized early (at RF or at an IF/baseband stage) and everything after that is math running on a processor, FPGA or PC. That math does the bandpass/notch filtering that crystal or mechanical filters used to do, performs detection (demodulating SSB, CW, FM, AM, or a digital mode), and on transmit builds the modulated waveform before it goes to the DAC. Because those steps are code rather than fixed hardware, a firmware update can add a new mode or change filter bandwidth continuously.
Watch out Picking only one function misses the point of SDR: filtering, detection, and modulation are all just DSP operations on sampled data, so no single one of them is the answer.
In an SDR the hardware just digitizes; software does filter, detect, and modulate.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C1212 of 14

What is the frequency above which a low-pass filter's output power is less than half the input power?

Why A filter's cutoff frequency is defined as the point where output power drops to half the input power, which is -3 dB in decibels (10 log 0.5 = -3 dB). For a low-pass filter, signals below cutoff pass with little loss, and above cutoff attenuation increases steadily. That -3 dB half-power point is the standard reference used to specify any filter's passband edge.
Watch out Rolloff sounds plausible but it describes how fast attenuation increases beyond cutoff (dB per octave or decade), not a single frequency; a notch frequency is the deep rejection point of a band-stop filter, and the neper is just a unit of attenuation ratio based on natural logs.
Cutoff = half power = -3 dB. Rolloff is the slope, cutoff is the spot.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C1313 of 14

What term specifies a filter's maximum ability to reject signals outside its passband?

Why Ultimate rejection is the maximum attenuation a filter can provide to signals well outside its passband, expressed in dB. Every real filter has a floor: stray coupling around the filter and component limitations mean a strong out-of-band signal eventually leaks through at some level, and that leakage level sets the ultimate rejection figure. A crystal or IF filter might be specified as, say, 80 dB ultimate rejection, meaning nothing outside the passband is attenuated more than that no matter how far removed in frequency.
Watch out Rolloff is the tempting one, but it describes how steeply the response falls off near the passband edges (dB per octave or per kHz), not the ultimate attenuation floor far out; insertion loss is the small attenuation the filter adds inside its passband, and notch depth applies to a single rejected frequency.
Ultimate rejection = the best the filter can ever do far out; rolloff = how fast it gets there.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
G7C1414 of 14

The bandwidth of a band-pass filter is measured between what two frequencies?

Why Filter bandwidth is defined between the two frequencies where the response has dropped to half the power of the peak, which is 3 dB down in voltage terms (0.707 of peak voltage). The span between that lower half-power point and the upper half-power point is the passband width. This same -3 dB convention defines bandwidth for resonant circuits, amplifiers, and receiver IF filters.
Watch out "Cutoff and rolloff" sounds plausible because cutoff frequency is a real term, but rolloff describes the slope of the skirts outside the passband, not a single defining frequency; the cutoff frequency itself is simply the half-power point.
Bandwidth = the -3 dB points. Half power, 0.707 voltage, top and bottom.
HamSandwich explanation, first draft. The question and answers are the NCVEC text.
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