A question was posed recently by Bill, N2CQR on his Soldersmoke Blog (link) about how to accomplish audio bandpass filtering for his homebrew DSB (Double SideBand) transmitter.
"Conventional" SSB transmitters limit the passband to about 2.1 kHz of audio over a range of about 300 Hz to 2.4 kHz and in so-doing, they cut off the lowest bass notes as well as the higher frequencies. In the nearly 150 years since the telephone was invented it has long been observed that human speech can be easily understood and recognized with this limitation and it's a good thing, too, since these early systems weren't capable of even that! While these extra frequencies are certainly "nice" to have - as any AM enthusiast will tell you - they aren't absolutely necessary for perfectly intelligible speech.
As it turns out - in the interest of best efficiency it's not necessary to transmit those lowest-frequency portions of human speech, anyway, since they carry relatively little information: It's largely the harmonics of these sounds and they way that they are articulated that convey the aspects of human speech that we need to understand one another. In fact, the voice of the typical adult human male has its fundamental energy almost entirely below the 300 Hz frequency range with the majority of the energy in the area around 1000-1500 Hz, plus or minus a bit. Additionally, the typical, small speaker found built into radios simply isn't capable of reproducing these low frequencies very well, anyway.
When SSB came along one of the several schemes for producing it involved the use of filtering to remove the opposite sideband as well as additionally-attenuate the carrier to be removed, and for reasons of practicality it was convenient to remove the low-frequency speech components while scraping off the highs above about 2.4 kHz. Doing this allowed more channels of audio to fit in multi-carrier transmissions systems (such as telephone cables) as well as on the amateur bands.
If the frequencies below 300 Hz or above 2400 Hz weren't transmitted, why receive them? The same type of filtering (and often the very same filter!) that used to filter the transmitted signal was also used to limit the passband of the received signal as well.
So, considering a simple, homebrew DSB transmitter, what would be the simplest way to shape the audio to produce the desired 300-2400 Hz audio passband for efficient, intelligible modulation?
An obvious, "old-school" scheme would be to use an entirely passive system consisting of chokes and capacitors but these days, high-inductance chokes of the values necessary to provide the desired passband at an impedance commensurate with a low-impedance (say, 1k) microphone aren't in the typical experimenter's junkbox or parts bins! (Or are they? See the next installment...)
Active bandpass filter with a 3rd order lowpass and 2nd order highpass.
Click on the image for a larger version.
The next, most obvious approach would be the "wide-band" bandpass filter which is essentially a low-pass filter and high-pass filter that are cascaded (the order isn't really important...) In order to get a nice, flat and relatively "steep" response at the low and high end and to keep the filter somewhat simple I chose a simple 3rd-order "Sallen-Key" circuit like that depicted in the "Active Filter Cookbook" by Don Lancaster over more complex designs: This filter is depicted in Figure 1, above.
Intentionally, the component values were made to be those that are commonly-available - particularly the resistors and capacitors, intentionally "re-using" as many of the same values as possible throughout. In this filter, R1/C1 provide RFI filtering on the input while U1 and associated components provide a low-impedance source for the input of the filter while minimally-loading the audio input which could be a high or low impedance microphone. U2 and associated components provide the two poles of lowpass filtering with the third pole inexpensively synthesized with C8 and R12, effectively rolling off the audio above about 2.5 kHz at the rate of 18dB/octave. U3 and associated components provide the highpass response, removing audio below about 300 Hz, forming a 2-pole filter. The final two components, C9 and R13 provide DC-blocking and minimum output impedance termination, respectively: Without R13, a capacitive load such as an RF bypass capacitor can cause instability of an op-amp based circuit. The final circuit consisting of U4 supplies a clean, low-impedance source at one-half of the power supply voltage to make the op amps happy! While LT1058 op amps were shown, this was selected only out of convenience in LTSpice (tm) and about any op amp - single, dual or quad - could be used instead. (Suggestions include a pair of NE5534's, a single TL074 or TL084, and an LM324 could be used in a pinch.)
The predicted response of the bandpass filter - nice and flat on the top.
Click on the image for a larger version.
- -2dB at 420 and 2300 Hz
- -6dB at 330 and 2900 Hz
- -20dB at 220 and 4000 Hz
- -40dB at 136 and 6300 Hz
I've not (yet) built this particular filter, but I've found that op-amp based filters are generally foolproof if one pays attention to detail when it comes to selecting the proper components values and agree nicely with the simulated values.
Admittedly, the circuit shown strays a bit from being "as simple as possible" - particularly if the intent, as is often the case with many QRP enthusiasts, is to not use any integrated circuits.
For a follow-up on this article, see the post "An L/C audio bandpass filter using cheap audio transformers" (link)
This page stolen from ka7oei.blogspot.com