[sdiy] How to use an analog time-domain multiplexer?

Benjamin Tremblay btremblay at me.com
Thu Jun 3 15:13:43 CEST 2021

I know the old copper telephone system used multiplexers like this for phone-grade audio. Bell Labs established the idea that you must have sample rate 2x the highest frequency to prevent hard distortion, and that applies to analog delay lines, too. So if you run the multiplexer so each channel gets 22kHz of air time, you will get distortion after 11KHz.
Lesson learned: as long as you constrain your solution to a spec, you’re going to hit the target, and if not, roll back your spec. 
The phone system had the advantage of only having to support the original telephone spec. So quiet, you can hear a pin drop, through a carbon microphone.

Personally I think you can get full audio spectrum, with some artifacts. The higher the clock frequency, the easier to hide the artifacts. 

But, let’s consider the possibility of using ADC/DAC and serial ports. Seems like a USB “sound card” (as the Asians call them) podcast mixer is competition for passing multiplexed audio over a single cable.

> On Jun 3, 2021, at 7:03 AM, cheater cheater via Synth-diy <synth-diy at synth-diy.org> wrote:
> Switches like the 4067 have the bandwidth to pull 16 audio channels
> through one wire, but it's not entirely clear to me how to use them
> correctly. I roughly know that I need to have a band limiting filter
> before the analog switch, then i need a level shifter / line driver,
> then the signal goes into a cable, then there's a receiving buffer /
> level shifter, then the other switch, and then possibly an s&h and
> reconstructing filter. However, I just don't know that much about how
> this should be done.
> For example, I don't really know much about the S&H / reconstructing
> filter part. Isn't the signal coming out of the switch basically a
> multiplication of two signals: (desired audio) * (gate), where the
> gate happens at 1/16 the switching frequency. Looking at data sheets,
> the Nexperia part seems to have a t_on and t_off of roughly 60 ms
> each, meaning it can switch at ~ 8.33 MHz; so one channel would be
> sampled at 520 kHz. Given that the channel only /contains/ 22 kHz
> audio, wouldn't (desired audio) * (gate) only contain two frequency
> bands? one's at 0-22 kHz (the desired audio) and an alias around 520
> kHz. 520/22 = 23.6.., So wouldn't reconstruction be as simple as
> putting down a simple 2nd-order filter at 25 kHz?
> Of course you don't want to run at 8 MHz if not necessary, so maybe
> the switching frequency could be lowered to 4 or 2 MHz, but a filter
> could still easily separate the desired audio with the aliases.
> Am I on the right track here?
> Regarding line drivers... what sort of problems could I run into here?
> I understand I'd need to look at the GBW of the op amp and make sure
> it fits what I'm doing, but other than that?
> I'd like to run this signal over a normal mini jack cable - so I'm not
> sure if this requires any special massaging.
> BTW, is it a good idea to run the analog switches using Gray code, or
> is sequential selection better?
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