Small Solar Chargers

George Ghio wrote:

meow2222@care2.com wrote: George Ghio wrote:
I did build a kit inverter, once, years ago. It had a max rating of 150W, Which it met.
It had a half hour rating of 0W
And a surge of about 300W
I'm still scratching my head over that one.
Which part don't you understand

I didnt seriously think you were going to stand by the above specs. Since you are, enough said.
NT

wmbjk wrote:

On 23 Dec 2005 23:57:27 -0800, meow2222@care2.com wrote:
George Ghio wrote:
I did build a kit inverter, once, years ago. It had a max rating of 150W, Which it met.
It had a half hour rating of 0W
And a surge of about 300W
I'm still scratching my head over that one.
NT
He previously wrote about using either a 150 Ohm rheostat (other times referred to as 150A or 200A), or 300k nichrome wire to control a few Amps of field current on a small 12V automotive alternator. According to the "designer" (who often refers to himself as a "solar power consultant"), 150 Ohms didn't allow sufficiently low output, hence the need for the nichrome wire. And as we all know, the prime consideration on a home-power generator is low output. <snorf> Similar wisdom surely underpins his other projects. For instance, liberal application of 300k wire might be useful when building a max 150, peak 300, zero Watt inverter, or most any zero Watt inverter for that matter. Perhaps strung around the property in large coils like razor wire at Slinky Manor. ;-)
Wayne

:) The trouble is he may actually BE a solar consultant. Nothing stopping him afaik. At least not for a few years anyway. :)
NT

daestrom wrote:

So, if I have a signal with a 1000 hz carrier, with a bandwidth of 50 hz, you think I can sample it at just 150 hz and get accurate reproduction? That's just wrong.
It is the maximum frequency component in the signal that is important. The bandwidth is not related unless the lower edge of the band is at 0 hz (whereupon the upper side of the band is equal to the max frequency).
daestrom
No, it is correct. If you have a signal with a 1000 Hz carrier and a 50

Hz Bandwidth, you can indeed sample it at 150 Hz and get accurate reproduction...provided the rest of the spectrum is clear. That requirement is typically provided with an anti-alias filter. In this case, the anti-alias filter has to be a BAND-PASS filter centered on 1000 Hz rather than the low pass filter associated with baseband sampling. This works because sampling folds the spectrum (aliasing) so that parts of the frequency band with higher frequency than the sampling frequency get folded back onto baseband. As long as the full spectrum only has energy in a bandwidth less than or equal to half the sample rate, you get all of the information necessary to reconstruct the original signal (assuming you know the characteristics of the fixed anti-alias filter so that you know which image to select when you unfold the spectrum). If there was signal energy outside of the Fs/2 bandwidth, it adds to signal inside the bandwidth during the folding that sampling causes, and then you lose information since there is no way to separate the energy if it has been added with other energy by folding.

SolarFlare wrote:

The point is the sampling rate has to be done at just over double the frequency of the signal and not the bandwidth.

No, that is not correct. It only needs to be sampled at > 2x the bandwidth, assuming the spectrum has been properly filtered to energy outside the signal of interest. See my earlier post.

SolarFlare wrote:

If only the baseband frequency is sampled at 6kHz then information is missing to recreate the original 100kHz and the sampling information is insufficient to recreate the original signal.
This is analogous to saying the number 1234 can be represented by (1234-234) / 1000 = 1
If I supply the number 1.0 you can regenerate the number 1234 from it? Not true, without the rest of the sampling information. The sample is incomplete.
Bandwidth sampling only cannot recreate the original signal.

You've used the wrong part of 1234 for your example. The proper analogy would be to say that 1234 can be represented by 234 in a 3 digit decimal number system. In that case, the overflow caused by exceeding 999 results in 1234 aliasing onto 234. If you know that all your input numbers are between 1000 and 1999, then 234 is sufficient information to represent 1234 with no ambiguity.
The anti-alias filter on your sampling system performs the bracketing to make sure that all the possible inputs are constrained to be within a bandwidth of your center frequency +/- BW/2, so when sampled there is no aliasing. In essence, that filter is the constraint that makes it work.
BTW, the same holds true for baseband sampling: The numbers in a baseband system based on your example are assumed to be less than 1000, so that 234 accurately represents 234. In that case if you put in 1234, it would also map to 234 and you'd have an ambiguity. It just so happens that in the baseband case, the representation is the same as the original signal for signals within the bandwidth allowed by Fs/2. With other than baseband, the representation is not the same as the number represented, but the constraints imposed by the system allow you to reconstruct the original value without ambiguity.

"Ray Andraka" wrote in message

daestrom wrote: So, if I have a signal with a 1000 hz carrier, with a bandwidth of 50 hz, you think I can sample it at just 150 hz and get accurate reproduction? That's just wrong.
It is the maximum frequency component in the signal that is important. The bandwidth is not related unless the lower edge of the band is at 0 hz (whereupon the upper side of the band is equal to the max frequency).
daestrom
No, it is correct. If you have a signal with a 1000 Hz carrier and a 50 Hz Bandwidth, you can indeed sample it at 150 Hz and get accurate reproduction...provided the rest of the spectrum is clear. That requirement is typically provided with an anti-alias filter. In this case, the anti-alias filter has to be a BAND-PASS filter centered on 1000 Hz rather than the low pass filter associated with baseband sampling. This works because sampling folds the spectrum (aliasing) so that parts of the frequency band with higher frequency than the sampling frequency get folded back onto baseband. As long as the full spectrum only has energy in a bandwidth less than or equal to half the sample rate, you get all of the information necessary to reconstruct the original signal (assuming you know the characteristics of the fixed anti-alias filter so that you know which image to select when you unfold the spectrum). If there was signal energy outside of the Fs/2 bandwidth, it adds to signal inside the bandwidth during the folding that sampling causes, and then you lose information since there is no way to separate the energy if it has been added with other energy by folding.

You are in effect demodulating the incoming signal and sampling the result, not sampling the incoming signal. You are 'throwing away' the information that would tell you what the carrier freq is.
Now, in radio that may be all well and good, since demodulation is a necessary part of reception anyway. But some of us were talking about reproducing the incoming signal, not stripping out the low freq component of some carrier.
Note that if the carrier is an exact multiple of the sample rate, *then* an unmodulated carrier will produce no 'alias' signal. But 150 doesn't go evenly into 1000.
If you have a completely unmodulated 1000 hz signal, passed through a 50 hz wide band-pass, centered around 1000 hz and sampled at 150 hz, your sampled data is indistinguisable from that of a 25 hz signal. Even knowing the band-pass filter's characteristic doesn't tell me if the carrier was unmodulated 1000 hz, or if there was a true 30 hz signal modulating it.
daestrom

Ah I see. Non compus mentis is your base state.
Let's See if we can clear up the story for you.
150W max continuous rating. Didn't think that would have been to hard to glean from the post if someone knew how inverters are rated.
0W half hour rating. That would indicate that the inverter in question does not, in fact, have a half hour rating.
300W surge.This means that the inverter will supply 300W for a couple of seconds or less. Enough to start a small motor or a TV.
Sorry you are TSP.
meow2222@care2.com wrote:

George Ghio wrote:
meow2222@care2.com wrote:
George Ghio wrote:
I did build a kit inverter, once, years ago. It had a max rating of 150W, Which it met.
It had a half hour rating of 0W
And a surge of about 300W
I'm still scratching my head over that one.
Which part don't you understand
I didnt seriously think you were going to stand by the above specs. Since you are, enough said.
NT

daestrom wrote:

"Ray Andraka" wrote in message
daestrom wrote:
So, if I have a signal with a 1000 hz carrier, with a bandwidth of 50 hz, you think I can sample it at just 150 hz and get accurate reproduction? That's just wrong.
It is the maximum frequency component in the signal that is important. The bandwidth is not related unless the lower edge of the band is at 0 hz (whereupon the upper side of the band is equal to the max frequency).
daestrom

No, it is correct. If you have a signal with a 1000 Hz carrier and a 50 Hz Bandwidth, you can indeed sample it at 150 Hz and get accurate reproduction...provided the rest of the spectrum is clear. That requirement is typically provided with an anti-alias filter. In this case, the anti-alias filter has to be a BAND-PASS filter centered on 1000 Hz rather than the low pass filter associated with baseband sampling. This works because sampling folds the spectrum (aliasing) so that parts of the frequency band with higher frequency than the sampling frequency get folded back onto baseband. As long as the full spectrum only has energy in a bandwidth less than or equal to half the sample rate, you get all of the information necessary to reconstruct the original signal (assuming you know the characteristics of the fixed anti-alias filter so that you know which image to select when you unfold the spectrum). If there was signal energy outside of the Fs/2 bandwidth, it adds to signal inside the bandwidth during the folding that sampling causes, and then you lose information since there is no way to separate the energy if it has been added with other energy by folding.
You are in effect demodulating the incoming signal and sampling the result, not sampling the incoming signal. You are 'throwing away' the information that would tell you what the carrier freq is.
Now, in radio that may be all well and good, since demodulation is a necessary part of reception anyway. But some of us were talking about reproducing the incoming signal, not stripping out the low freq component of some carrier.
Note that if the carrier is an exact multiple of the sample rate, *then* an unmodulated carrier will produce no 'alias' signal. But 150 doesn't go evenly into 1000.
If you have a completely unmodulated 1000 hz signal, passed through a 50 hz wide band-pass, centered around 1000 hz and sampled at 150 hz, your sampled data is indistinguisable from that of a 25 hz signal. Even knowing the band-pass filter's characteristic doesn't tell me if the carrier was unmodulated 1000 hz, or if there was a true 30 hz signal modulating it.
daestrom

The information that tells you the frequency of the carrier is not discarded, but is partially implied by the system, just as it is with a baseband system. Remember, sampling is essentially the mixing of the signal with an impulse train, followed by a sample rate decimation without any filtering. The choice of frequencies in this example are unfortunate because there is in fact some interference between the positive and negative frequency images of the original signal. When dealing with real-only inputs, you need to be judicious in selecting the sample frequency so that the frequency folding does not fold the negative image (that is a reflection of the positive image and is always present for a real signal) onto the positive image. Still, that doesn't mean that the sample frequency has to be a sub-multiple of the carrier. For example, 160 Hz sampling works (as does 210 Hz) with a 1000Hz signal that has a 50Hz bandwidth because it puts both the positive and negative frequency images into the sampled spectrum without overlap. There is sufficient information there to reconstruct the original signal if the center frequency of the anti-alias filter is known.
And yes, you are correct that the sampled signal is indistinguishable from one which it aliases onto: but those other frequencies are not present in the signal thanks to the anti-alias filter. The point is that the anti-alias filter needn't be a low pass filter. It can be a band pass filter as long as the bandwidth is less than half the sample frequency. If the input signal is a real signal, there are additional considerations to make sure that the postive and negative frequeyncy images do not overlap when the spectrum is folded.

George Ghio wrote:

Ah I see. Non compus mentis is your base state.
Let's See if we can clear up the story for you.
150W max continuous rating. Didn't think that would have been to hard to glean from the post if someone knew how inverters are rated.
0W half hour rating. That would indicate that the inverter in question does not, in fact, have a half hour rating.
300W surge.This means that the inverter will supply 300W for a couple of seconds or less. Enough to start a small motor or a TV.
Sorry you are TSP.
meow2222@care2.com wrote:
George Ghio wrote:
meow2222@care2.com wrote:
George Ghio wrote:

I did build a kit inverter, once, years ago. It had a max rating of 150W, Which it met.
It had a half hour rating of 0W
And a surge of about 300W

I'm still scratching my head over that one.

Which part don't you understand

I didnt seriously think you were going to stand by the above specs. Since you are, enough said.
NT

"Ray Andraka" wrote in message

daestrom wrote: "Ray Andraka" wrote in message
daestrom wrote:
So, if I have a signal with a 1000 hz carrier, with a bandwidth of 50 hz, you think I can sample it at just 150 hz and get accurate reproduction? That's just wrong.
It is the maximum frequency component in the signal that is important. The bandwidth is not related unless the lower edge of the band is at 0 hz (whereupon the upper side of the band is equal to the max frequency).
daestrom

No, it is correct. If you have a signal with a 1000 Hz carrier and a 50 Hz Bandwidth, you can indeed sample it at 150 Hz and get accurate reproduction...provided the rest of the spectrum is clear. That requirement is typically provided with an anti-alias filter. In this case, the anti-alias filter has to be a BAND-PASS filter centered on 1000 Hz rather than the low pass filter associated with baseband sampling. This works because sampling folds the spectrum (aliasing) so that parts of the frequency band with higher frequency than the sampling frequency get folded back onto baseband. As long as the full spectrum only has energy in a bandwidth less than or equal to half the sample rate, you get all of the information necessary to reconstruct the original signal (assuming you know the characteristics of the fixed anti-alias filter so that you know which image to select when you unfold the spectrum). If there was signal energy outside of the Fs/2 bandwidth, it adds to signal inside the bandwidth during the folding that sampling causes, and then you lose information since there is no way to separate the energy if it has been added with other energy by folding.
You are in effect demodulating the incoming signal and sampling the result, not sampling the incoming signal. You are 'throwing away' the information that would tell you what the carrier freq is.
Now, in radio that may be all well and good, since demodulation is a necessary part of reception anyway. But some of us were talking about reproducing the incoming signal, not stripping out the low freq component of some carrier.
Note that if the carrier is an exact multiple of the sample rate, *then* an unmodulated carrier will produce no 'alias' signal. But 150 doesn't go evenly into 1000.
If you have a completely unmodulated 1000 hz signal, passed through a 50 hz wide band-pass, centered around 1000 hz and sampled at 150 hz, your sampled data is indistinguisable from that of a 25 hz signal. Even knowing the band-pass filter's characteristic doesn't tell me if the carrier was unmodulated 1000 hz, or if there was a true 30 hz signal modulating it.
daestrom

The information that tells you the frequency of the carrier is not discarded, but is partially implied by the system, just as it is with a baseband system. Remember, sampling is essentially the mixing of the signal with an impulse train, followed by a sample rate decimation without any filtering. The choice of frequencies in this example are unfortunate because there is in fact some interference between the positive and negative frequency images of the original signal.

Actually, I kind of chose those numbers for that very reason ;-)
When

dealing with real-only inputs, you need to be judicious in selecting the sample frequency so that the frequency folding does not fold the negative image (that is a reflection of the positive image and is always present for a real signal) onto the positive image. Still, that doesn't mean that the sample frequency has to be a sub-multiple of the carrier. For example, 160 Hz sampling works (as does 210 Hz) with a 1000Hz signal that has a 50Hz bandwidth because it puts both the positive and negative frequency images into the sampled spectrum without overlap. There is sufficient information there to reconstruct the original signal if the center frequency of the anti-alias filter is known.
And yes, you are correct that the sampled signal is indistinguishable from one which it aliases onto: but those other frequencies are not present in the signal thanks to the anti-alias filter. The point is that the anti-alias filter needn't be a low pass filter. It can be a band pass filter as long as the bandwidth is less than half the sample frequency. If the input signal is a real signal, there are additional considerations to make sure that the postive and negative frequeyncy images do not overlap when the spectrum is folded.

So what you're saying is, *if* you know the carrier frequency and band-width of the signal imposed on that carrier, you can design a system that will be able to reproduce the imposed signal using a relatively low sample rate (low when compared to the carrier frequency).
But if the carrier frequency changes, then you need to modify the sample rate to avoid a lot of aliasing issues. So in radio reception, the sample rate is adjusted along with tuning the receiver? Or is this done at the intermediate frequency which is fixed so that sample rate adjustment is fixed with the intermediate frequency? (do they even still use superheterodyning in tuners?? ;-)
It's been a long time since I did any RF stuff. But A/D and D/A stuff at AF and lower has been quite a passion for me for some time. And the basic Nyquist hasn't changed.
daestrom

"daestrom" wrote in message

So what you're saying is, *if* you know the carrier frequency and band-width of the signal imposed on that carrier, you can design a system that will be able to reproduce the imposed signal using a relatively low sample rate (low when compared to the carrier frequency).

It's a litle more general than that -- you only need to know that your signal lies inbetween some lower and upper frequencies and that bandwidth is (generally) less than 1/2 of the sample rate of the ADC.

But if the carrier frequency changes, then you need to modify the sample rate to avoid a lot of aliasing issues.

Assuming all the "information" (the carrier and whatever sideband(s) you care about) is still within your bandpass frequencies, you've lost nothing and there is no aliasing with any non-zero signals.

So in radio reception, the sample rate is adjusted along with tuning the receiver?

Not usually, although there are so many ways to build 'a radio,' I'm sure this approach has been implemented at some point in time.
It pretty common to digitize significantly more of a radio band than the bandwidth of the signal you're interested in and then just digitally track & demodulate the one signal you need from the many that are present. This is popular because none of the 'fundamental' settings of the system (local oscillator frequencies, IF frequencies, ADC sample rate, anti-alias filters, etc.) change; this makes the architecture inexpensive and highly flexible. The downside is that sensitivity can be poor if there are other, stronger sides in the band that you've digitized but aren't really interested in... A common fix for this problem is to stick an adjustable notch filter somewhere in the analog path, but of course that adds cost again... etc, etc, etc... we sit around all day making these tradeoffs. :-) Another common fix is to switch to frequency hopping spread spectrum modulation like Bluetooth uses. (From a certain point of view, people like the cell phone carriers have it easy in that they _own_ the spectrum they're operating in and know _exactly_ what signals should be present, their power levels, etc. -- That makes their radio designs noticeably simpler and cheaper than "general purpose" wideband receivers that are used by, e.g., the military, hams, etc.)

Or is this done at the intermediate frequency which is fixed so that sample rate adjustment is fixed with the intermediate frequency?

This is quite common.

(do they even still use superheterodyning in tuners?? ;-)

Superheterodyning is still common to get the RF down to an IF that can be digitized directly. As Ray mentioned earlier, the problem with trying to digitize, say, a narrowband 900MHz signal using a 5Msps ADC is that the effect of any clock jitter going into the ADC gets multiplied by the 900/5, so at some point obtaining a decent oscillator becomes impractically expensive.
---Joel

OK let's go with your analogy example of 1234 being represnted by 234 only.
You have no way of decoding 234 into 1234 without passing information of 1000 as your baseband info and therefore the the number 1234 has not been successfuly representedm as being reproduced without further information.
Now we could further argue algorythms as part of the information or part of the sample.
"Ray Andraka" wrote in message

You've used the wrong part of 1234 for your example. The proper analogy would be to say that 1234 can be represented by 234 in a 3 digit decimal number system. In that case, the overflow caused by exceeding 999 results in 1234 aliasing onto 234. If you know that all your input numbers are between 1000 and 1999, then 234 is sufficient information to represent 1234 with no ambiguity.

Your sampling of only the bandwidth information is not complete for decoding.
"Ray Andraka" wrote in message

SolarFlare wrote:
The point is the sampling rate has to be done at just over double the frequency of the signal and not the bandwidth.

No, that is not correct. It only needs to be sampled at > 2x the bandwidth, assuming the spectrum has been properly filtered to energy outside the signal of interest. See my earlier post.

Now you have to provide samples ***AND*** changing information with an algorthm to decode successfully. Your samples are then not complete and useless without other information supplied.
Superhetrodyning in a radio assumes a variable superhetrodyning frequency when it gets decoded
Let's see you regenerate the original carrier and information from that without the carrier frequency known.
When you listen to the audio on your radio can you tell the carrier frequency without the dial?
"Joel Kolstad" wrote in message

Assuming all the "information" (the carrier and whatever sideband(s) you care about) is still within your bandpass frequencies, you've lost nothing and there is no aliasing with any non-zero signals.
Superheterodyning is still common to get the RF down to an IF that can be digitized directly. As Ray mentioned earlier, the problem with trying to digitize, say, a narrowband 900MHz signal using a 5Msps ADC is that the effect of any clock jitter going into the ADC gets multiplied by the 900/5, so at some point obtaining a decent oscillator becomes impractically expensive.
---Joel

"SolarFlare" wrote in message

Now you have to provide samples ***AND*** changing information with an algorthm to decode successfully. Your samples are then not complete and useless without other information supplied.

This is true, but it's pretty much true for all communication systems, not just sub-sampled digital ones. What changes is that sometimes the extra 'information' supplied can be done by something as sophisticated as a human's brain as he tunes across the dial to find the 'best' sound -- this corresponding to finding the carrier. (At some point this becomes a very philosophical discussion... 'information' only has _meaning_ to an observer who presumably knows something about or has a hunch as to what they're observing is. Although one can compute the 'information' within signal in an attempt to ascertain whether it resembles a random process or whether it's conveying what we may 'intelligence.' Hence you can probably recognize the difference between someone speaking random jibberish and an actual language even without knowing that language, but on the other hand a good way to conceal information is to make it appear almost completely random -- when in fact it isn't --, which is exactly what cryptography does.)

Let's see you regenerate the original carrier and information from that without the carrier frequency known.

Example: Take an antenna that's about a meter long... feed its output to an LNA and then a reasonably steep bandpass filter passing 144-148MHz... sample with a 16 bit, 12MSps ADC (I chose 12 just because it shifts 144MHz to baseband, although there's no reason you can't use any frequency >8Msps). Feed this digital word to a 16 bit DAC clocked at 10MSps. Lowpass filter the DAC's output with a reasonably steep 4MHz low-pass filter. Feed this signal to one port of a mixer and 144MHz to the other port. Poof! There's your original signal back again! Feed this through another 144-148MHz bandpass filter if you don't like the image response at 140-144MHz.
There are a few caveats here:
1) Clock jitter will tend to broaden out the specctra of the original signals a bit (how good is your clock?) 2) The track & hold (analog) circuitry in the ADC has to be good to a couple hundred MHz to avoid distortion. 3) Your noise floor is limited to no better than ~-100dB (and potentially _much_ worse if you haven't been careful in your layout, power supply decoupling, etc.). Note that everything described above also applies to switched capacitor circuits (a technology whose time has just about passed, but a neat idea); in that case analog noise rather than quantization noise will dictate the noise floor (and realistically it'll probably be much worse than -100dB...) 4) A typical DAC holds its output (e.g., a first-order hold) rather than generating impulses, so the spectrum reproduced has a sin(x)/x profile to it (frequencies closer to 148MHz will have less gain than those at 144MHz); this can be fixed in the digital domain with the use of a FIR or IIR filter. In some systems the droop is small enough that people just ignore it.
This example is reasonably practical. Strictly speaking, to make it simlper you can just bandpass filter the output of the DAC directly and be OK, but the sin(x) profile along with the limited analog bandwidth of the DAC tend to make this approach impractical (you end up with very little SNR); this approach if often used for proof-of-concept demos, though.
If you happen to have a carrier at 146.23MHz in the input signal, it'll most certainly still be there in the output signal, yet the system didn't have to 'know' where the carrier was.

When you listen to the audio on your radio can you tell the carrier frequency without the dial?

Sure, I can measure it! In fact, a much more interesting problem is how one generates a carrier when none exists in the first place. There are plenty of modulation schemes out there specifically don't use a carrier to either save power (TV transmissions -- which have a very small albeit not eliminated carrier -- are a good example of this) or to conceal transmissions (in military systems nothing attracts unwanted attention more than a carrier some 60dB above the noise floor).
---Joel