# Help for multiplexer design

Discussion in 'General Electronics Chat' started by Elekta, Jan 19, 2010.

1. ### Elekta Thread Starter New Member

Jan 19, 2010
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0
Hi all

I would like to design an analog 8 audio channels multiplexer ( 8 to 1 ), and I have a couple of questions please. I'm wondered if anybody cna help me.

The audio bandwidth of each channel should be from 10Hz to 25 kHz.
In what manner do I have to calculate the MUX ( clock ) fequency please ?
Is it correct:

fmux = fmax channel * 2 * channel number ?

( moltiplication by 2 is for Nyquist )

On my case:
fmux = 25kHz * 8 * 2 = 400 kHz ?

Or can I just consider 25kHz * 8 = 200 kHz ?

How can I sincronize Muplitplexer with the demultiplexer in order to extract each single ghannel please ? ( 1 to 8 )

Thank you very much for your help

2. ### beenthere Retired Moderator

Apr 20, 2004
15,815
282
Going from the standard for digitizing audio, a multiplexing frequency of 22.1KHz is good. That would suggest that 176.8KHz would be enough to multiplex 8 audio streams.

Can't answer the synchronizing question - you have given no details about the project. You may be able to use the same clock, or have to send some additional data to do the decoding.

3. ### Elekta Thread Starter New Member

Jan 19, 2010
6
0
Thank you very much Beenthere.

So, if I understood correctly, I need just to multiply 22k1 ( bandwidth of each channel ) * number of channels ? Is it not necessary to consider ( 22k1*2 ) * 8 ?

About the project. It is very very simple I think.

I have to take 8 analog sources ( 8 microphones ) and to multipex them into one channel.

Then I would like to send this MUX via R.F.

In the receiver, I ned to deMUX it.
It serves to avoid 8 balanced cables from the mic station to the place in which the audio will be recorded.

4. ### beenthere Retired Moderator

Apr 20, 2004
15,815
282
It gets more interesting if you are doing stereo, as the sampling rate per channel doubles. One sample is the right channel, the other is the left.

As normal digitizing is a continuous alternation of R -L, R - L, etc, imposing a wait for the sampling of the additional 7 channels means there will be some dead time at the normal sample rate, possibly leading to loss of signal quality. I imagine you will be gating the analog signals through the multiplexer. Each segment will be a period of the signal, followed by 7 equal periods of silence. That seems to mean a loss of 7/8ths of the total signal.

You may need to try out the scheme to see if the sampling rate will need to be considerably increased.

5. ### Elekta Thread Starter New Member

Jan 19, 2010
6
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Well I'm not interested to stereo broadcast... I need to transfer 8 mic's signale froma station to another just multiplexing ... So I have not the problems related to bandwidth and so on ... but I need to know:

* how to calculate the clock frequency for the multiplexer ... in order to avoid aliasings

* how to sync the tarnsmeitter with the receiver. In this case I can use the clock directly ... transmitting it ...

Do you think it's a good idea ? I have not so many other choices I saw :-/

6. ### beenthere Retired Moderator

Apr 20, 2004
15,815
282
Unless you are digitizing, aliasing is not the problem. Keeping the demultiplexing in synch with the multiplexer is. Adding an additional signal at a frequency that may be separated out of the audio feed with a filter may be the way to do it. You only need to signal which is channel #1, although sending the multiplexing clock avoids other problems.

I would want to try this scheme on a smaller scale to see what problems might arise from chopping the audio streams.

7. ### Elekta Thread Starter New Member

Jan 19, 2010
6
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Well, I'm not digitizing anything ... in other words: I will just send analog signals to analog multiplexer ... as 2 * CD4046 for instance ... and a shift register that will selcet one by one the channels ...

How to tell to the decoder that's teh chanel 1 the other the 2 and so on ... sincerely talking: I have not clue ... the first thing I'm thinking is: in Stereo, nobody tells to the receiver when channel L is sent and when the R. I know ha the 19 kHz pilot is taking in phase the local oscillator ( 38 kHz ) and in this way it decodes correctly ... so that I need a PLL circuitin hte receiver to to this.

But in th transmitter, I have then two choices:

1 - to send the carrier as it is ... but it will create some problems because the lowest audio frequency is 10-20 Hz .... it means: I have not enough guard-band from the carrier to filter it out ... and at the same time every filtering operation changes the phase of the carrier increasing cross-talk level ... too bad ...

2 - to send a multiple of the carrier ... so high that it's out of audio range and its spectrum. In this way I can avoid filtering diminishing phase distorsion.

Another manner, I was thinking about one hour ago, is to "copy" in what manner VIDEO system is sending sync ... and to do it also with audio ... I mean:

between the audio packet of each channel, a peak is sent ... look the attachement to have more clear idea about what I'm telling, please ....

But those are just iedeas ...

• ###### MUX.GIF
File size:
31.6 KB
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Last edited: Jan 19, 2010
8. ### beenthere Retired Moderator

Apr 20, 2004
15,815
282
That might work very well. You could place two pulses between channel 8 and channel 1 to identify the channel order.

9. ### Elekta Thread Starter New Member

Jan 19, 2010
6
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Indeed. I will be wondered f there are already multiplexers and demultiplexers that are doing all those things ... I mean that over the MUXed line there is already sync signal and while they are deMUXing automatically recognize all things ... it will make my life extremely easier ... I will just realize the analogue circuitry to interface microphones to the MUX and the remaining operations are done by the chip ... if you know the existence of those chips that are doing everything please don't hesitate to tell me

10. ### Elekta Thread Starter New Member

Jan 19, 2010
6
0
I think to have found a solution, in order to synch the MUX and the DEMUX. Of course is on paper but I not found any other indication.

I was thinkinking to send a burst in superaudio ( I mean something over the 22k1 ... I tell around 30 kHz ) when the channel 1 appears.

To the DEMUX site, it detects che burst at 30 kHz and automatically resets the counter making start the DEMUX from channel 1 too.

What do you think ?

I'm looking for very simple solutions.

Thank you very much