RNEW 320 Exam 3 Student Name 1. Please read the following descriptions and write down your reactions. ‘What Does the Spectrum Analyzer Measure B.1. Introduction The oscilloscope is great for observing more or less repetitive signals that are constant in time, but it gives only scant idea of the frequency content of any but the simplest of signals. The spectrum analyzer, on the other hand, gives little information on the time- variations, but a superbly detailed picture of the frequency content. Both are powerful and necessary tools for designing and debugging photo-measurement systems. The spectrum analyzer is used for noise and signal-to-noise measurements, for choosing modulation frequencies, for tracking down spurious signals and interferences, Indeed, there is a lot of merit in investigating all completed designs with a good spectrum analyzer, in order to see problems with spurious emissions and weak or out-of-band instabilities that go unnoticed on a time-domain display. This investigation must extend well beyond the notional signal frequency, as high-frequency instabilities can have a marked and difficult to understand effect on signal performance. However, there is a lot of confusion about what the spectrum analyzer actually shows, which has been compounded by the new generation of digital analyzers, PC data~ acquisition systems, and fast fourier transform (FFT) computing oscilloscopes. Hence it is useful to work though a few simple examples in pedantic detail (the simple ones we all get wrong!). B.2. 1V Peak Sine Wave Ifa +1V peak (2V pk-pk) sine-wave voltage at 1kHz (Fig. B.1) is applied to a conventional spectrum analyzer, what does the main spectral line read on the ordinate scale? To answer that we need to make a small detour into power definitions, and this means "dB.” dBs describe ratios, so when using them we must always say what is the power to which the value is referenced. For historical reasons, in radio electronics the basis of the scale is usually defined as a root mean-square (rms) power of 1mW dissipated in a resistance of 500 (Take care. Figure B.1 One volt peak sine wave.a70r ms Voltage in $0.Ohm Time (rs) The audio engineers often use 6000, but we will ignore them for now). This is defined as OdB referred to ImW, or OdBm. This is the basis of our scale of power ratios. One milliwatt in 500 also implies a voltage of 0.2236V rms, or 0.2236 * 2V2 = 0.6324V plepk our +1 peak sine-wave voltage, with an rms voltage 1/2 = 0.7071, generates a power of 0.7071 2/50 = calculated from the power ratio: OmW rms. In dBm this is 101og,,(10mW/1 mW) = 10dBm. As the vertical axis of the spectrum analyzer is usually calibrated in dBm, this is the magnitude that should appear at the peak of the 1kHz line. The displayed power in dBm can alternatively be calculated directly from the voltage as: 20log 1.(0.7071V/0.2236V) = 10dBm Note the term "20," as we have now calculated a ratio of voltages, not powers. It is a pity for much of modem laboratory electronics that the measurements are so usually phrased in terms of powers (which are rarely measured directly), in rms voltages (which are much harder to read on an oscilloscope than peak-to-peak voltages) and in matched 509 source/detector impedances (when the majority of measurement are made at the low-impedance outputs of opamps and voltage references detected in a high impedance scope probe). Nevertheless, with a little care the pitfalls can be avoided,Table B.1 Synthesis of a 1V peak Square-wave Prem oncray 1 0.900 16.21 3 0.300 1.80 5 0.180 0.65 7 0.129 0.33 9 0.100 0.20 " . 0.082 0.13 19.3 B.3. 1V Peak Square wave As ajust slightly more complex example, connect a 1V peak 1kHz square wave to the ‘spectrum analyzer. (Just check first that it can handle the power dissipation). The 1kHz fundamental should read about +12dBm. How do we know that this is right? Well, it is straightforward to perform a Fourier analysis of the square wave to determine the coefficients, that is, the amplitudes of each of the harmonics that make up the square wave. This can be done analytically and the result approximates our original square wave as: 4/r [sin x + 1/3 sin 3x + 1/5 sin 5x + ...] These coefficients 4/n, 4/3n, 4/5r etc. of each harmonic are the peak voltages of the sine waves that make up the symmetrical square wave. | don't think Fourier mentioned rms values, which are of course ¥2 times lower, and must be used to calculate the power in dBm delivered by each harmonic. This is done in Table B. Note that the sum of the powers delivered by the first six component sine waves is 19.3mW, close to the 20mW we expect for a 1V peak square wave or a continuous 1V in 500, Figure B,2 shows the first three sine waves with the peak values from column two, their sum, and the original square wave that made them. There is generally no true zero- frequency component on the spectrum analyzer's display. Hence we can offset the square wave on top of a DC voltage without changing the display, as long as it is not overloaded. B.A, Spectrum Calculation With the wide availability of digital oscilloscopes, RS232, USB and wireless connection to your PC, and drag-and-drop waveform display and analysis, it is very attractive to avoid paying for an expensive spectrum analyzer. Why not just store a long time trace and calculate the spectra in software? It certainly is a useful approach, but just as in understanding the units of a spectrum analyzer's display, similar care must be applied when we use the discrete Fourier transform (DFT) to calculate spectra. One problem is ‘that there are many different definitions of these DFTs, usually calculated using the fast Fourier transformFigure B.2 One volt peak square wave decomposed into the first three harmonic components, 14 ta 1.0 08 E 06 5 04 B02 £ 00 3° 3 aan 20m in 50 Ohm 02 04 06 08 10 Time (ms) Figure B.3 Mathcad printout. Sampled sine wave and discrete FFT result. WHAT DOES THE FFT MEASURE? Number of samples: N:=128, N re) Plothe me tuncton: —y, in| 2%) Perform the Fower transform (calculates thepeak voltage): = 2am) Figure B.4 Mathcad printout. Sampled square wave and discrete FFT result.Nowa square wave: mits x, Plot the time function: v,2=¥,- 1 0 S10. Nm Perform the Fourier tanstorm, and display the frequency domai:RNEW 320 3rd Midterm — Studlant rame___ S preblens, 20% each ee "oA tees 65 lasted. Tasks, Find DO = Find Q, = 7 Find b, =? Note. « tt) x os 20a avs fo Der Za, cs ttre + Fb siuttty2. Please read the following descriptions and write down your reactions. Standard Filter Types Filters And Their Uses You may well have encountered visual filters in Photoshop or social media platforms like Instagram. Such filters manipulate the composition of an image removing colors, saturation and shades. They can also increase such content Just as a water filter, separates unwanted fine particles from a purified result, audio passes through an audio filter and unwanted content is removed Just like image filters, but unlike most other types of filter, audio filters can also amplify elements of what is present in the source audio, to produce a new desired audio result. In reality filters are frequency dependent circuits, with audio filters working within the audio frequency range, 0 Hz to 20 kHz. There are many different filter types. Audio filters are used to amplify (boost), pass or attenuate (cut) source signals at selected frequency ranges. Filters are used for many different audio applications. These include instrument amplifiers, effects units, musical synthesizers, recording studios, sound reinforcement, hi-fi systems, computer audio and virtual reality set ups In practical terms filters in an EQ are used to treat issues, like enhancing a vocal sound, or treating drum recordings to reduce spill or bringing clarity to a muddy mix by reducing unwanted bass and low mid frequencies and avoiding instruments cancelling each other out. The filters used in synthesizers are related, though controls and circuitry may differ. Many of the goals remain the same. In this article we will focus on the EQ applications of filters, High and Low Pass Filters High Pass Filters and Low Pass Filters are the most type of audio filter. These filters begin to cut the amplitude of an input signal at a selected threshold frequency or cutoff frequency. In the case of a high pass filter, all frequencies below a selected frequency will be progressively cut. Similarly, for a low pass filter, the frequencies above a selected threshold frequency will be progressively cut. These filters are sometimes called Cut-off Fitters.High and Low Pass filters commonly have a gentle cut of -6 dB per octave, -12 dB per octave or a more aggressive -18 dB per octave. The circuits used to implement filters are not ideal. This means that the gain does not instantly change from 0 dB to -18 dB as the frequency crosses the threshold. The gain change is progressive. The quoted cut off frequency is the frequency at which the gain reduction is already 3dB below the input amplitude. This point is also called the 34B Down Point High Pass Filter High Pass Filter Parametric EQ’s generally provide a High Pass Filter (HPF) An HPF removes frequencies below a frequency set by a frequency control. Frequency is cut from below the set frequency at a steep curve, Frequencies above that frequency remain unaffected. In other words it passes the high frequencies and cuts the low frequencies, hence the two names for this filter. High Pass Filters are ideal for removing any unwanted low-end frequencies, such as mains hum and other low frequency noise. They are also commonly used to clear space for the bass and kick drum in a mix.Low Pass Filter Low Pass Filter Parametric EQ's generally also provide a Low Pass Filter (LPF) An LPF removes frequencies above a frequency set by a frequency control, Frequency is cut from above the set frequency at a steep curve. Frequencies below that frequency remain unaffected. In other words it passes the low frequencies and cuts the high frequencies, hence the two names for this filter. Low Pass Filters are ideal for removing any unwanted high-end frequencies, such as fan noise, hiss and other high frequency noise. They are also commonly used to clear space for cymbals in a mix.Bell Filters or Peak Filters Bell Filters ar Peak Filters Bell curve filters are also known as peak filters. A bell curve attenuates or boosts frequencies around a specified center frequency. The bandwidth Q control sets the width of the bell curve, or the breadth of the frequencies affected. Bell curves are very versatile filters. They can be used to boost or reduce a range of frequencies, They can be very accurate. For example, Set a low Q for a broad curve, and gentle gain able to subtly boost a wide range of musical frequencies. Set a high Q for precise narrow cuts to correct any troublesome signal Shelving Filters This kind of fiter applies an equal level of boost or cut to all frequencies beyond a user defined threshold frequency. Yet again the filter response is not ‘idealHigh Shelf Filter High Shelf Filter A high shelf filter will attenuate or boost all frequencies above a specified frequency, adjusted by the frequency control, to the level adjusted by the gain control. Unlike low pass filters/ high cut filters, high shelf filters don’t cut frequencies out completely Instead, they reduce or boost high frequencies to a plateau gain level. Shel filters make broad tonal changes by boosting or reducing high-end frequencies.Low Shelf Filter Low Shelf Filter A low sheff filter will attenuate or boost all frequencies below a specified frequency, adjusted by the frequency control, to the level adjusted by the gain control. Unlike high pass filters/ low cut filters, low shelf filters don’t cut frequencies out completely. Instead, they reduce or boost bass frequencies to a plateau gain level. Shelf filters make broad tonal changes by boosting or reducing low-end frequenciesBand-Pass Filters Band-Pass Filter A Band-Pass Filter passes frequencies, in a chosen passband around a specified frequency. Frequencies beyond the band width are not allowed through the filter. Essentially beyond the Q (bypass and) all frequencies are cut (attenuated) A high-@ filter has a narrow passband while a low-0 filter has a wide passband. A high- Q filter is also known as a narrow-band band-pass filter. A low-Q filter is also known as a wide-band band-pass filter. Stop-Band Filter A Band-Stop Filter or Band-Rejection Filter, is the opposite of a Band-Pass Filer. Stop- Band Filters cut (attenuate) all frequencies, within a chosen Q (stopband), around a specified frequency. As with a band-pass passband, a high-Q filter has a narrow stopband while a low- Q filter has a wide passband. A high-Q filter is also known as a narrow-band notch filter. A low-Q filter is also known as a wide-band notch filterNotch Filter Notch Filter A Notch Filter is a narrow-band Band-Stop Filter. They are also called a "T-Notch Filter’, “Band Limit Filter’, “Band-Elimination Filter’, and “Band-Reject Filter’ The stopband is typically 1 to 2 decades wide i.e. the highest frequency cut is 10 to 100 times the lowest frequency cut, when used outside of audio applications. For audio applications, notch filters have high and low frequencies that can be just semitones apart https://www.songstuff.com/recording/article/standard_filter_types/S37 ohms .E ry 537 ohms t=? Cine costes) eo We = 2 Centeaff angular fragnenty ) algae ( Trans far function es ea ic) ase rr tn 3e10€ a b= “ 27 ohm ae : Fe Sin na toe ig ae = an peye) Garrats Urge a Sint sae We =? C Critical anguler foaguancy ) aus? 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