## Hypothetical schema for sampling maxima and minima of a continuous analog signal.

I have just examined a hypothetical application of modern IC technology. What’s already available out-of-the-box is, that for every Analog – Digital Conversion, a single value can be generated. And today’s chips allow this to happen at 100Msps, resulting in a bandwidth of 50MHz. But for some reason, the goal might be desired, that for each of these conversions, a pair of values is generated, that accurately reflect what the maximum and minimum analog value was. As a result, a hypothetical ~oscilloscope~ could display a column of pixels, with one X-coordinate, instead of just one pixel, indicating one Y-coordinate. And so, this is the first of 3 diagrams I came up with:

My assumption is, that a 100Msps A/D Conversion will follow each of the above circuit’s 2 outputs…

(Updated 5/27/2021, 10h30… )

## Successive Approximation

While Successive Approximation is generally an accurate approach to Analog-to-Digital conversion, it is not a panacea. Its main flaw is in the fact that the D/A converter within, will eventually show inconsistencies. When that happens, some of the least-significant bits output will either be an overestimated one, followed by nothing but zeroes, or an underestimated zero, followed by nothing but ones.

Although circuit specialists do what they can to make this device consistent, there are quantitative limits to how successful they can be. And, whether 24 bits can be achieved depends mainly on frequency. In analog circuits, voltages tend to zero in on an ideal voltage exponentially, even when there is no signal-processing taking place. So the real question should be, ‘Can 24 bits still be achieved, far above 48kHz?’

And, if we insist that the low-pass filter should be purely numeric, we are also implying that one A/D conversion must be taking place at the highest sample-rate, such as at 192kHz, while if the low-pass filter could be partially analog, this would not be required.

Dirk

## About The History of Sinc Filters

A habit of mine which betrays my age, is to use the term ‘Sinc Filter’. I think that according to terminology today, there is no such thing. But there does exist a continuous function called ‘the Sinc Function’.

When I use the term ‘Sinc Filter’, I am referring to a convolution – a linear filter – the discreet coefficients of which are derived from the Sinc Function. But I think that a need exists to explain why such filters were ever used.

The Audio CDs that are by now outdated, were also the beginning of popular digital sound. And as such, CD players needed to have a Digital-to-Analog converter, a D/A converter. But even back when Audio CDs were first invented, listeners would not have been satisfied to listen to the rectangular wave-patterns that would come out of the D/A converter itself, directly at the 44.1 kHz sample-rate of the CD. Instead, those wave-patterns needed to be put through a low-pass filter, which also acted to smooth the rectangular wave-pattern.

But there was a problem endemic to these early Audio CDs. In order to minimize the number of bits that they would need to store, Electronic Engineers decided that Human Hearing stopped after 20 kHz, so that they chose their sampling rate to be just greater than twice that frequency. And indeed, when the sample-rate is 44.1 kHz, the Nyquist Frequency, the highest that can be recorded, is exactly equal to 22.05 kHz.

What this meant in practice, was that the low-pass filters used needed to have an extremely sharp cutoff-curve, effectively passing 20 kHz, but blocking anything higher than 22.05 kHz. With analog circuits, this was next to impossible to achieve, without also destroying the sound quality. And so here Electronics Experts first invented the concept of ‘Oversampling’.

Simply put, Oversampling in the early days meant that each analog sample from an D/A converter would be repeated several times – such as 4 times – and then passed through a more complex filter, which was implemented at first on an Analog IC.

This analog IC had a CCD delay-line, and at each point in the delay-line it had the IC equivalent to ‘a potentiometer setting’, that ‘stored’ the corresponding coefficient of the linear filter to be implemented. The products of the delayed signal with these settings on the IC, were summed with an analog amplifier – on the same IC.

Because the Sinc Function defines a brick-wall, low-pass filter, ifÂ  a 4x oversampling factor was used, then this linear filter would also have a cutoff-frequency at 1/4 the new, oversampled Nyquist Frequency.

What this accomplished, was to allow an analog filter to follow, which had 2 octaves of frequency-separation, within which to pass the lower frequency, but to block this oversampled, Nyquist Frequency.

Now, there is a key point to this which Electronics Experts were aware of, but which the googly-eyed buyers of CD players were often not. This type of filtering was needed more, before the Analog-to-Digital conversion took place, when CDs were mastered, than it needed to take place in the actual players that consumers bought.

The reason was a known phenomenon, by which If a signal is fed to a sample-and-hold circuit running at 44.1 kHz, and if the analog, input frequency exceeded the Nyquist Frequency, these excessive input frequencies get mirrored by the sample-and-hold circuit, so that where the input frequencies continued to increase, the frequencies in the digitized stream would be reflected back down – to somewhere below the Nyquist Frquency.

And what this meant was, that if there was any analog input at an supposedly-inaudible 28.05 kHz for example, it would wind up in the digital stream at a very audible 16.05 kHz. And then, having an oversampling CD player would no longer be able to separate that from any intended signal content actually at 16.05 kHz.

Therefore, in studios where CDs were mastered, it was necessary to have the sample-and-hold circuit also run at 4x or 8x the final sample-rate, so that this could be put through a homologous low-pass filter, only 1/4 or 1/8 the samples of which would actually be converted to digital, through the A/D converter, and then stored…

Now today, that sort of filter design has been replaced completely, through the availability of better chips, that do all the processing numerically and therefore digitally. Hence, if 4x oversampling is being used, the digital version of the signal and not its analog version, are being ‘filtered’, through specialized digital chips.

Back in the 1980s, the types of chips and the scale of integration required, were not yet available.