I have to disagree with you on these points. You once mentioned several DSP books you are using in your courses. My recollection is that they all explain this reasonably well.... I had thought of using impulse function but then it wasn't making sense to me in the given context.
... An impulse train can't produce such a spectrum.
... I have checked some books but no one explains Nyquist theorem the way it should be explained.
I have to disagree with you on these points. You once mentioned several DSP books you are using in your courses. My recollection is that they all explain this reasonably well.
Yes, exactly. I was just saying that this explanation is found in many of the books you have access to. Notice how short the descriptions are. Just one short paragraph is needed because they are using very powerful sweeping logic to take a difficult question and strip it down to its essential elements. When viewed from this vantage point, the answer should appear simple if you have a solid understanding of the basic system theory tools used.But I'm sure that you are just saying that many of those e-books provide a fair explanation of sampling. I agree with you on this.
The answer to this is simple if you trust all the system theory you've learned.A single impulse contains theoretically all the frequencies with equal magnitude. But how come a train of impulses only consist of train of frequencies and not the continuum of frequencies.
OK, the final sampled signal looks like a train of impulse with unequal amplitudes. You are bringing up the actual physical process by which we might make the final sampled signal.In real sampling, a high-speed switch is turned on for only the small period of time when the sampling occurs. The result is a sequence of samples that retains the shape of the analog signal. The most common sampling method is sample and hold. I believe in this context the operation of turning on the switch for very small period of time is denoted by an impulse. The "f_s" denotes the number of samples taken per second or number of impulses generated per second. (This is where I got stuck yesterday and then asked you about that impulse train with unequal amplitudes). As a matter of fact when switch is turned on for a very small period, small capacitor is charged up. So, turning on of the switch and charging of the capacitor is what sampling is all about, in simple words. So, where is that impulse train which we intend to use for the sampling? I think that turning on of the switch and charging of the capacitor makes up that impulse train. The switch and capacitor mechanism would give us that impulse train where all the impulses have equal amplitude only if the capacitor gets charged up to the same level every time but the capacitor doesn't get charged up to equal amount every time.
Here, I think bringing up electrons and how they vibrate is completely off base. System theory does not need to consider the physical mechanisms which compelled us to use the system theory. All that matters is whether the physical system approximates the many critical assumption required by our system theory.For the sake of continuity of this discussion, I'm going to assume that we have an impulse train with equal amplitudes. Suppose the analog signal is a sine wave with frequency of 5 Hz. It would mean that we will get two spikes in frequency domain for such an analog signal where one spike occurs at 5 Hz and the other at -5 Hz. Further assume that the impulse train gives spikes at 10 Hz, 20 Hz, 30 Hz and so on; the spikes also extends in negative direction like this -10 Hz, -20 Hz, -30 Hz, and so on. I will only focus on positive frequencies for simplicity. The analog signal having frequency of 5 Hz means that the electrons are acting like microscopic mass-spring systems with frequency of 5 Hz. Likewise, when it comes to impulse train, it means that some electrons are acting like mass-spring systems with frequency of 10 Hz, others with 20 Hz and so on. Now comes another point which is really confusing me. How can two of such mass-spring systems interact that their frequencies get added up? For a mass-spring system angular frequency, omega, is sqrt(k/m).
The text says that we can filter out the higher harmonics and can get the DC value of 3V. I believe the higher harmonics have less amplitude, i.e. volts, than the DC component but they still contain energy. So, don't you think eliminating them or shorting them out using the capacitor result is waste of energy? Thank you.
Regards
PG
So, don't you think eliminating them or shorting them out using the capacitor result is waste of energy?
Usually, filters operate on signals which have very little energy, so wasting energy is not an issue.
Thank you, NG, nsaspook.
I still remember that a pulse or rectangular wave could only have a DC component if it's symmetric around the y-axis. I might be wrong so please bear with me and correct me. In my view, the 1st waveform will have a DC component while the 2nd one won't. Is this correct?
Which of the two waveforms is used in PWM analysis?
PID control would be appropriate if you decided to implement an outer loop with P&O maximization control, and an inner loop controlling the input voltage could be done with PID.
Personally, I would implement a dual feedback with P&O providing control as an outer slow loop, and a faster loop controlling the input voltage (input resistance could be done too, but is more complicated ). The outer loop would command input voltage, and the inner loop would command duty cycle. By doing this, you allow the slow outer loop to control for sun/cell variation, and the inner loop controls against load/temperature changes in the buck converter itself. This might be overkill (then again, it might not be depending on the actual application), but it's not difficult and typically would not add more cost because a slow P&O algorithm does not strain even the most basic microprocessor, and hence leaves processing power for a faster inner loop.
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