The Simplest Possible Mixer, using MOSFETs.

When a curious person searches the Internet for the circuit diagrams of (electronic) mixers, there is a certain complexity of what he or she will find. Just for people who might not know, the type of mixer I’m referring to is a component which does not add two signals together – which is what the naming might seem to suggest – but rather, which multiplies two signals. In certain cases the mixer will produce output, that contains an additive component as well as a multiplied component. But it’s the multiplied component circuit designers are interested in, because that can be used:

  1. In order to produce ‘mixed frequencies’, between two input frequencies, such as between a local oscillator and a Radio Frequency, resulting in an Intermediate Frequency,
  2. In order to act as a phase discriminator, the output of which will be maximally positive or negative, when two input signals are in-phase, but the output-voltage of which will be some neutral voltage, when the input waves are 90⁰ out-of-phase with each other. In this latter case, two reasonably constant input amplitudes are assumed.

What search results will often show, is somewhat complex mixers, that require either one or two balanced inputs – meaning inputs conditioned such, that they each appear differentially between two input electrodes – and which have as advantage for being designed that way, low distortion of the wave-form(s) supplied differentially in this way.

But sometimes, low distortion is not required. For example, in the case of a PLL – a “Phase-Locked Loop” – It’s assumed that the feedback voltage changes the frequency of a VCO – a “Voltage-Controlled Oscillator” – but with the intended result that two outputs lock in some phase-position, so that the two frequencies that are inputs to the phase-discriminator will be exactly the same frequency. This latter need often arises in the design of ICs. This latter application does not require that the phase-discriminator be particularly linear, nor that its output-voltages, that become feedback voltages, be in any range other than the range which the VCO requires as input.

And so the question can arise, what the simplest way might be to design a mixer, with the added detail that both inputs are unbalanced inputs – i.e., that each input appears at one terminal, and not in an opposing way, at two terminals – and for the sake of argument, our IC might be limited to using enhancement-mode, N-channel MOSFETs as the main active component. And this would be my solution:

Coinc-Det_1.svg

The concept is very simple. If Vin1 and Vin2 are at 180⁰, then M1 and M2 don’t conduct simultaneously. Therefore, R1 and Vcc (the supply voltage) achieve maximally positive average output-voltage. If Vin1 and Vin2 are at 0⁰ phase-position, the two transistors will become conductive in a way that coincides. Therefore, this is actually a Coincidence Detector. And the average  output-voltage will be maximally negative in that case. And, if Vin1 and Vin2 are at a 90⁰ phase-position, then the average output-voltage will be somewhere between the two values mentioned before.

I suppose it should be mentioned that, if the circuit designer knows ahead of time that one of the two inputs has a much higher amplitude than the other, or a more predictable amplitude, then this usually stronger input should be fed to Vin1.

As part of a feedback loop, the output needs to be followed by a low-pass filter, that emulates an integrator over the time-constant which is the fastest, with which that feedback loop is supposed to be able to react to a change in one of the frequencies. The simplest low-pass filter consists of a resistor followed by a capacitor… (:1)

And so, when looking for a way to implement a phase-discriminator, the curious person needs to choose which of the following has greater priority:

  • The simplest circuit-design, or
  • The lowest amount of distortion.

The circuit above will certainly give the highest amount of distortion. :-P

(Updated 7/9/2019, 16h55 … )

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NG-SPICE: Low-Powered Saw-Wave Generator

The goal of my latest exercise at using the Open-Source circuit simulation software named ‘NG-SPICE’ consisted of designing a low-powered saw-wave generator. Here were the premises of the project:

  • A train of pulses is to be taken as input, that are approximately of 1μS duration, 2V in amplitude, and that have a steady rate of recurrence of 100kHz.
  • They are to be converted into a saw-wave that has an attack as fast as the pulses are short, and which has approximately linear falloff after each input pulse.
  • One active component is a monolithic N-channel enhancement-mode MOSFET transistor with a gate size of approximately 100 microns squared – which therefore has poor qualities if compared to discrete components – but which is plausible as part of an IC with Medium Scale Integration (:2)
  • The other active component is a bipolar diode of unknown weaknesses, which has been approximated as a discrete 1N4148 switching diode.
  • The entire circuit is to operate off a 3V power supply.
  • The maximum output load is in the vicinity of 100kΩ – 40kΩ, and must not change the internal workings of this circuit block. (:1)
  • The output amplitude is to reach approximately +1V with respect to the circuit ground.

What was observed:

  • The diodes were difficult to get into a conductive state at the low pulse-voltage.
  • The chosen MOSFET makes a very poor output driver.

Saw_1

Screenshot_20190628_131425


 

The experiment seems to have been successful.

(Updated 7/3/2019, 8h35 : )

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A Basic Limitation in Stereo FM Reproduction

One of the concepts which exist in modern, high-definition sound, is that Human Sound perception can take place between 20 Hz and 20kHz, even though those endpoints are somewhat arbitrary. Some people cannot hear frequencies as high as 20kHz, especially older people, or anybody who just does not have good hearing. Healthy, young children and teenagers can typically hear that entire frequency range.

But, way back when FM radio was invented, sound engineers had flawed data about what frequencies Humans can hear. It was given to them as data to work with that Humans can only hear frequencies from 30Hz to 15kHz. And so, even though Their communications authorities had the ability to assign frequencies somewhat arbitrarily, they did so in a way that was based on such data. (:1)

For that reason, the playback of FM Stereo today, using household receivers, is still limited to an audio frequency range from 30Hz to 15kHz. Even very expensive receivers will not be able to reproduce sound, that was once part of the modulated input, outside this frequency range, although other reference points can be applied, to try to gauge how good the sound quality is.

There is one artifact of this initial standard which was sometimes apparent in early receivers. Stereo FM has a pilot frequency at 19kHz, which a receiver needs to lock an internal oscillator to, but in such a way that the internal oscillator runs at 38kHz, but such that this internal oscillator can be used to demodulate the stereo part of the sound. Because the pilot signal which is actually part of the broadcast signal is ‘only’ at 19kHz, this gives an additional reason to cut off the audible signal at ‘only’ 15Khz; the pilot is not meant to be heard. But, way back in the 1970s and earlier, Electrical Engineers did not have the type of low-pass filters available to them which they do now, that are also known as ‘brick-wall filters’, or filters that attenuate frequencies above the cutoff frequency very suddenly. Instead, equipment designed to be manufactured in the 1970s and earlier, would only use low-pass filters with gradual ‘roll-off’ curves, to attenuate the higher frequencies progressively more, above the cutoff frequency by an increasing distance, but in a way that was gentle. And in fact, even today the result seems to be, that gentler roll-off of the higher frequencies, results in better sound, when the quality is measured in other ways than just the frequency range, such as, when sound quality is measured for how good the temporal resolution, of very short pulses, of high-frequency sound is.

Generally, very sharp spectral resolution results in worse temporal resolution, and this is a negative side effect of some examples of modern sound technology.

But then sometimes, when listeners with high-end receivers in the 1970s and before, who had very good hearing, were tuned in to an FM Stereo Signal, they could actually hear some residual amount of the 19kHz pilot signal, which was never a part of the original broadcast audio. That was sometimes still audible, just because the low-pass filter that defined 15kHz as the upper cut-off frequency, was admitting the 19kHz component to a partial degree.

One technical accomplishment that has been possible since the 1970s however, in consumer electronics, was an analog ‘notch filter’, which seemed to suppress one exact frequency – or almost so – and such a notch filter could be calibrated to suppress 19kHz specifically.

Modern electronics makes possible such things as analog low-pass filters with a more-sudden frequency-cut-off, digital filters, etc. So it’s improbable today, that even listeners whose hearing would be good enough, would still be receiving this 19kHz sound-component to their headphones. In fact, the sound today is likely to seem ‘washed out’, simply because of too many transistors being fit on one chip. And when I just bought an AM/FM Radio in recent days, I did not even try the included ear-buds at first, because I have better headphones. When I did try the included ear-buds, their sound-quality was worse than that, when using my own, valued headphones. I’d say the included ear-buds did not seem to reproduce frequencies above 10kHz at all. My noise-cancelling headphones clearly continue to do so.

One claim which should be approached with extreme skepticism would be, that the sound which a listener seemed to be getting from an FM Tuner, was as good as sound that he was also obtaining from his Vinyl Turntable. AFAIK, the only way in which this would be possible would be, if he was using an extremely poor turntable to begin with.

What has happened however, is that audibility curves have been accepted – since the 1980s – that state the upper limit of Human hearing as 20kHz, and that all manner of audio equipment designed since then takes this into consideration. This would include Audio CD Players, some forms of compressed sound, etc. What some people will claim in a way that strikes me as credible however, is that the frequency-response of the HQ turntables was as good, as that of Audio CDs was. And the main reason I’ll believe that is the fact that Quadraphonic LPs were sold at some point, which had a sub-carrier for each stereo channel, that differentiated that stereo channel front-to-back. This sub-carrier was actually phase-modulated. But in order for Quadraphonic LPs to have worked at all, their actual frequency response need to go as high as  40kHz. And phase-modulation was chosen because this form of modulation is particularly immune to the various types of distortion which an LP would insert, when playing back frequencies as high as 40kHz.

About Digital FM:

(Updated 7/3/2019, 22h15 … )

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