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Why we do that in AM demodulation?

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idmond

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Hey all,

"In AM synchronous demodulation, Why we don't divide m(t)coswt by cos(wt)
instead of multiplying by cos(wt), since this can be easily implemented by a simple divider circuit?"



If it's all about thinking mathematically, then it seems like it's more intuitive and a whole lot easier if we just divide by coswt, why we go through all the trouble and multiply then we have to know the trig identity of (cos(a)cos(b)) and then put a LPF after the output...

I asked this question to two professors and I got different answers:

#Professor 1 Reply:
"For your scheme to work you must know exactly what the frequency w is that the transmitter is using, which tends to drift. So try your scheme with dividing by cos (w+delta)t and see whether you can recover the signal."


#Professor 2 reply was:
"we don't use the division scheme for two reasons:

a- In the real world, noise is added to the received signal and it is going to look like this, m(t)coswt+n(t). If you then divide by coswt, you will get, m(t)+n(t)/coswt, and since the cosine function ranges from -1 to 1, thus for values of cosine less than 1, the noise term will be amplified instead of being attenuated , so you will get poor SNR.

b- When the cosine function goes to zero, you will divide by zero and this will cause amplitude spikes which leads to circuit saturation."



I am now confused more than ever:confused: which one is true?
 
Professor three says it is much harder to build a divider than a multiplier...
 
Professor three says it is much harder to build a divider than a multiplier...

that's a good one:). but why it's such a hard thing to do, because I always get this answer whenever I ask someone.
is it because that there are no actually divider circuits that can divide by a function(not a number like arithmetic dividers) or is it because that we really could implement a divider circuit that can divide by a sinusoid function like cosine, but it’s much easier to implement a multiplier circuit than a divider one.
 
none gave you complete answer but all professors are right
 
Cut the maths. Amplfy the AM signal to clipping point and use a simple 4066 analogue switch as a sampler. Feed the AM signal through this and use the clipped signal as a sampling signal on the gates of the switch. Filter the output and you have a synchronous demodulated AM signal.

Sampling is a simple multiplation process. You are sampling the signal at the envelope of the modulating frequency. Often overlooked by academics who have never worked in an industrial environment.
 
Cut the maths. Amplfy the AM signal to clipping point and use a simple 4066 analogue switch as a sampler. Feed the AM signal through this and use the clipped signal as a sampling signal on the gates of the switch. Filter the output and you have a synchronous demodulated AM signal.

Sampling is a simple multiplation process. You are sampling the signal at the envelope of the modulating frequency. Often overlooked by academics who have never worked in an industrial environment.

the clipped AM signal would be at the carrier frequency, so the received AM signal would be sampled at the clipped AM signal frequency, i,e the carrier frequency, right?

but don't you think that this method would be sensitive to any frequency deviation of the carrier of the AM signal since this carrier is used to control the sampling frequency, then I think that any deviation in the sampling instants would cause distortion in the output signal?
 
frequency drift is usually very slow, longer than the period of any recovered audio signal. so any errors from such a frequency drift will be very small. let's say the "Y" signal is generated by clipping the incoming carrier, and the "X" signal is the unmodified received signal. if X changes frequency, Y does as well at the same rate. so the sample rate is always the same as the received frequency of X. phase shifts of the X signal in a tuned circuit may cause minor errors with sudden shifts (as in "sudden" being within a time period of less than one cycle of the carrier), but such shifts are extremely rare. in the case of using a PLL to regenerate the received frequency, there will be delays introduced by the time constant of the loop filter, which will cause larger errors if the frequency shifts suddenly, as the VCO catches up with the input.
 
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