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AM digital demodulation methods

Started by gretzteam April 23, 2010
On Apr 24, 1:53&#4294967295;am, "gretzteam" <gretzteam@n_o_s_p_a_m.yahoo.com>
wrote:
> Hi, > I'm having trouble understanding the similarities/differences/advantages of > various AM digital demodulation methods, which makes me incapable of > deciding what to use. I would like to go throug a few methods here and here > your thoughts. > > First, I assume the common goal of all those techniques is to shift the > carrier back down to DC and there are a few ways to do this that have > different behavior. Right? > > Method 1) > -Bandpass filter around the carrier. > -Take the absolute value. THIS is what shifts the carrier down to DC. > -Low-pass filter. > Now I guess taking the absolute value is not a very good way to shift the > carrier down to DC and is sensitive to anything other signal in the stream. > This is why we need to bandpass? > Is the performance of the system pretty much only dominated by the bandpass > filter? In other words, is taking the abs() value messing up my data EVEN > if there was only the carrier/data in the stream? > > Method 2) > -Multiply incoming stream with sin(wc*t) and cos(wc*t) sample by sample. > -Lowpass filter each output. This gives I and Q. > -Calculate sqrt(I^2+Q^2). > > Here, the carrier is perfectly shifted down to DC no matter what the stream > contains, which is why we don't need the bandpass filter. Can we say that > the performance of such a system is only dependent on the lowpass filter? > What about the precision of the sin(wc*t) and cos(wc*t)? Keeping only one > bit here would mean using a square wave, which can't be really good. Is > there any way to know how much this affects performance? > > And probably most importantly, how can I measure performance of the system? > Is there a standard way. I'm not yet looking for complicated theoretical > explanations...something intuitive is better to learn at first - at least > for me! > > Thanks!
Synchronous demodulation using a PLL will give you 3dB improvement over ordinary envelope detection. The problem arrises when you want to do synchronous demod and the carrier isn't there! What I mean by that is that when you have double sideband supressed carrier. There is no power at the carrier freq then and nothing to lock onto. Solution...among otehr things you need to square the received waveform and lock into twice the carrier then divide down (missing some other crucial steps). Hardy
Clay <clay@claysturner.com> wrote:
(snip)
 
> This is a wonderful example of how a trivial method in analog (diode > and RC filter) is not really how you want to do it in digital. Now if > he has an analytic signal ....
I was almost going to mention that square wave demodulation is commonly used for the usual laboratory "lock-in amplifier", but again that is analog. The higher harmonics are normally just filtered out, but maybe not in the digital world. -- glen

HardySpicer wrote:


> Synchronous demodulation using a PLL will give you 3dB improvement > over ordinary envelope detection.
This is wrong. Synchronous demodulation makes improvement from 0dB to infinity, depending on SNR.
> The problem arrises when you want to do synchronous demod and the > carrier isn't there! > What I mean by that is that when you have > double sideband supressed carrier. There is no power at the carrier > freq then and nothing to lock onto. > > Solution...among otehr things you need to square the received waveform > and lock into twice the carrier then divide down (missing some other > crucial steps).
Let the dumb lead the blind. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
Jerry Avins wrote:
> On 4/23/2010 1:52 PM, gretzteam wrote: >>>> Use it, but understand it. Understand the implication of in-band >>>> interference. Understand the need to exclude out-of-band signals from >>>> the demodulation process. (The baseband low-pass filter can't remove >>>> aliases.) >>> >>> I am assuming that he is properly prepping the signal prior to the >>> multiplication by sin/cos and will pick appropriate filters at >>> baseband. >> >> >> Ok I must admit that I'm more confused than before! Why do you still >> need a >> bandpass filter for method 2? Isn't multiplying by sin/cos shifting the >> carrier frequency to DC? > > What Brent said. Keep in mind that you not only shift the carrier to > baseband, you also shift everything else down by a similar amount. Where > do the aliases of the out-of-band signals go? > >> About method 1 having the problem of peak values not being close to full >> scale, can we say that this is not a problem when fs>> carrier? > > When the carrier is adequately oversampled, method 1 works. I leave it > to you to determine what "adequate" means. How many samples per carrier > cycle are needed to ensure that one is at least 95% of either peak? Is > that a reasonable expenditure of resources?
Except that by his original description he's not peak-seeking -- he's averaging the absolute value. That _ought_ to work better, but I don't know by how much. -- Tim Wescott Control system and signal processing consulting www.wescottdesign.com

Clay wrote:


> This is a wonderful example of how a trivial method in analog (diode > and RC filter) is not really how you want to do it in digital. Now if > he has an analytic signal ....
Except for the diode detector doesn't work like the simplified textbook desccription. An accurate analysis requires involved math. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
>Synchronous demodulation using a PLL will give you 3dB improvement >over ordinary envelope detection.
3dB improvement of what? I don't quite understand how to qualify such a system. I assume you mean SNR, but how does it get measured?
>What Brent said. Keep in mind that you not only shift the carrier to >baseband, you also shift everything else down by a similar amount. Where >do the aliases of the out-of-band signals go?
Ok this is the part I don't understand! Can you elaborate a bit more? Thanks a lot!
On 4/23/2010 4:24 PM, Tim Wescott wrote:
> Jerry Avins wrote: >> On 4/23/2010 1:52 PM, gretzteam wrote: >>>>> Use it, but understand it. Understand the implication of in-band >>>>> interference. Understand the need to exclude out-of-band signals from >>>>> the demodulation process. (The baseband low-pass filter can't remove >>>>> aliases.) >>>> >>>> I am assuming that he is properly prepping the signal prior to the >>>> multiplication by sin/cos and will pick appropriate filters at >>>> baseband. >>> >>> >>> Ok I must admit that I'm more confused than before! Why do you still >>> need a >>> bandpass filter for method 2? Isn't multiplying by sin/cos shifting the >>> carrier frequency to DC? >> >> What Brent said. Keep in mind that you not only shift the carrier to >> baseband, you also shift everything else down by a similar amount. >> Where do the aliases of the out-of-band signals go? >> >>> About method 1 having the problem of peak values not being close to full >>> scale, can we say that this is not a problem when fs>> carrier? >> >> When the carrier is adequately oversampled, method 1 works. I leave it >> to you to determine what "adequate" means. How many samples per >> carrier cycle are needed to ensure that one is at least 95% of either >> peak? Is that a reasonable expenditure of resources? > > Except that by his original description he's not peak-seeking -- he's > averaging the absolute value. That _ought_ to work better, but I don't > know by how much.
How does the average value of samples of the carrier relate to the approximate value of the envelope? Would it help if the "carrier" were triangular? Jerry -- "It does me no injury for my neighbor to say there are 20 gods, or no God. It neither picks my pocket nor breaks my leg." Thomas Jefferson to the Virginia House of Delegates in 1776. &#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;
On 4/23/2010 4:32 PM, Vladimir Vassilevsky wrote:
> > > Clay wrote: > > >> This is a wonderful example of how a trivial method in analog (diode >> and RC filter) is not really how you want to do it in digital. Now if >> he has an analytic signal .... > > Except for the diode detector doesn't work like the simplified textbook > desccription. An accurate analysis requires involved math.
Except for diagonal clipping with deep modulation and the limited charging current, what is missing from the classical analysis? Whenever it was in my control, I used a full-wave peak detector. That suppresses the IF in the output, leaving the weaker second harmonic to dominate. You then need only a much smaller capacitor, making the onset of diagonal clipping more remote. Besides, 456 KHz can get through many Hi-Fi power amps, but the response at 902 is much less. Jerry -- "It does me no injury for my neighbor to say there are 20 gods, or no God. It neither picks my pocket nor breaks my leg." Thomas Jefferson to the Virginia House of Delegates in 1776. &#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;&#4294967295;
> > >How does the average value of samples of the carrier relate to the >approximate value of the envelope? Would it help if the "carrier" were >triangular? >
Yes you have a point here! All I've proven so far is that when the input signal contains only a carrier, full scale, then the output of the lowpass filter is pretty much exactly 0.63 (2/pi), which is the average value of a full scale sine wave. I was pretty happy to see this, but that's probably not AM demodulation just yet! But isn't this what the Analog version does when using bandpass, full wave rectifier and capacitor? Thanks

Jerry Avins wrote:

> On 4/23/2010 4:32 PM, Vladimir Vassilevsky wrote: > >> >> >> Clay wrote: >> >> >>> This is a wonderful example of how a trivial method in analog (diode >>> and RC filter) is not really how you want to do it in digital. Now if >>> he has an analytic signal .... >> >> >> Except for the diode detector doesn't work like the simplified textbook >> desccription. An accurate analysis requires involved math. > > Except for diagonal clipping with deep modulation and the limited > charging current, what is missing from the classical analysis?
Even with ideal diode and ideal source, the angle of conduction is determined by transcendental equation. Now account for impedance, nonlinearity and add noise, and it gets really messy.
> Whenever it was in my control, I used a full-wave peak detector. That > suppresses the IF in the output, leaving the weaker second harmonic to > dominate. You then need only a much smaller capacitor, making the onset > of diagonal clipping more remote. Besides, 456 KHz can get through many > Hi-Fi power amps, but the response at 902 is much less.
Jerry, do you know what was the rationale for choosing 455kHz vs 465kHz standard IF ? VLV