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From: Vladimir Vassilevsky on 23 Apr 2010 17:56 Jerry Avins wrote: > On 4/23/2010 5:33 PM, HardySpicer wrote: > >> On Apr 24, 8:24 am, Vladimir Vassilevsky<nos...(a)nowhere.com> wrote: >> >>> HardySpicer wrote: >>> >>>> Synchronous demodulation using a PLL will give you 3dB improvement >>>> over ordinary envelope detection. >>> >>> >>> This is wrong. >>> >> It's in the textbooks...read it! > > > What is ordinary envelope detection? Peak detection? Doesn't matter; It is very simple. Think of |I| vs sqrt(I^2 + Q^2) VLV
From: glen herrmannsfeldt on 23 Apr 2010 17:48 HardySpicer <gyansorova(a)gmail.com> wrote: (snip) > 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). As I understand it, commonly used by many modems doing anything except FSK. To make sure that there are enough transitions for carrier recovery, scramblers are commonly used on the bit stream. -- glen
From: gretzteam on 23 Apr 2010 18:09 >On 4/23/2010 5:03 PM, gretzteam wrote: >>> >>> >>> 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. > >How many samples per carrier cycle do you have? How many carrier cycles >do you average over? How long does that take, and what does that imply >about the highest envelope frequency you can demodulate without attenuation? Ok I'm way oversampled. I'm doing this to learn about it so I don't want to have the added difficulty of sample rate (just yet). Here is the current system - I should have posted this FIRST! parameters: fs = 4MHz carrier: 99kHz Currently, there is no noise, and no 'information' being modulated. Just a carrier sine wave:) One gotta start somewhere! I then bandpass using a 2nd order bandpass filter centered at the carrier. Then take the absolute value. Then lowpass filter using a 2nd order CIC filter all the way down to something ridiculous like 10-50Hz. The output matches surprisingly well the 2*A/pi formula depending on the A of the carrier. Now if I do a frequency sweep, using a full scale sine wave from 0 to 2MHz (fs/2), and plot the obtained average value after it settled, I get the shape of the bandpass filter! I guess this was to be expected, which is why I asked if 'method-1' was only dependent on the performance of the bandpass filter. I guess so far my 'information' is only at DC, but it works well. Now, ff I modulate a 2Hz signal, I also see it at the output, with a DC offset. And this is where I decided to post, since I didn't know how to measure performance of the system when there IS information. Does this make sense? Thanks!
From: Tim Wescott on 23 Apr 2010 19:21 gretzteam wrote: >> >> 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? Some analog implementations rectify and average, which is what you're doing. Some (the ones that have a series diode to a shunt cap) detect the peak of the waveform, then have the charge drained out of the cap by other circuit elements. I.e. a peak detector. Different things, but close enough when the carrier frequency is way higher than the audio. -- Tim Wescott Control system and signal processing consulting www.wescottdesign.com
From: Tim Wescott on 23 Apr 2010 19:23
Jerry Avins wrote: > 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? Well, the RF signal (not the carrier) is carrier * (audio signal + offset) -- so you can find a scaled value of the audio signal either by finding the peaks (as in traditional AM receivers) or by rectifying and averaging. I suspect (but would have to play with it to find out) that the rectify and average is not as harshly nonlinear, and therefore would stand a lower sampling (or carrier) rate. -- Tim Wescott Control system and signal processing consulting www.wescottdesign.com |