VRMS | Volunteer Relationship Management System
kandi X-RAY | VRMS Summary
kandi X-RAY | VRMS Summary
Volunteer Relationship Management System: This is an ambitious project to create a system that will help us measure our human capital development, reduce repetitive tasks and processes, and improve outcomes.
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QUESTION
I have a json file in which I store a dictionary of values. I know how to modify the value of a key individually, but I want to know how to update the dictionary within the json file with another dictionary.
json file called 'dummy.json'
...ANSWER
Answered 2021-May-27 at 07:48code:
QUESTION
I'm using MS Visual C++ 6.0 (as my work requires) on a pretty decent Windows 10 PC.
I'm having trouble with displaying the value after the decimal that is entered manually via "cin >>", I've tried the fix that was answered thru DSKVP's question (Show two digits after decimal point in c++) but it still won't show the decimal values, it just drops the digits after the decimal and/or rounds them off.
...ANSWER
Answered 2021-Mar-01 at 08:40I tested your code, and the only thing I changed was the return type of readResults()
from int
to double
. Once I did that, I got the results I expected!
QUESTION
I need to implement an RMS calculations of sine wave in MCU (microcontroller, resource constrained). MCU lacks FPU (floating point unit), so I would prefer to stay in integer realm. Captures are discrete via 10 bit ADC.
Looking for a solution, I've found this great solution here by Edgar Bonet: https://stackoverflow.com/a/28812301/8264292
Seems like it completely suits my needs. But I have some questions.
- Input are mains 230 VAC, 50 Hz. It's transformed & offset by hardware means to become 0-1V (peak to peak) sine wave which I can capture with ADC getting 0-1023 readings. Hardware are calibrated so that 260 VRMS (i.e. about -368:+368 peak to peak) input becomes 0-1V peak output. How can I "restore" back original wave RMS value providing I want to stay in integer realm too? Units can vary, mV will do fine also. My first guess was subtracting 512 from the input sample (DC offset) and later doing this "magic" shift as in Edgar Bonet answer. But I've realized it's wrong because DC offset aren't fixed. Instead it's biased to start from 0V. I.e. 130 VAC input would produce 0-500 mV peak to peak output (not 250-750 mV which would've worked so far). With real RMS to subtract the DC offset I need to subtract squared sum of samples from the sum of squares. Like in this formula:
So I've ended up with following function:
...ANSWER
Answered 2020-May-11 at 14:34I did not test your code, but it looks to me like it should work fine. Personally, I would not have implemented the function this way. I would instead have removed the DC part of the signal before trying to compute the RMS value. The DC part can be estimated by sending the raw signal through a low pass filter. In pseudo-code this would be
QUESTION
I'm trying to create a component based on react-table using composition so that we can use a customised version everywhere without repeating boiler plate. I'm using Typescript while react-table doesn't use it. I don't see any examples around of doing this so here's what I've come up with so far:
...ANSWER
Answered 2018-Oct-18 at 19:42QUESTION
I am trying to create a program in GNU Octave to draw a histogram showing the fundamental and harmonics of a modified sinewave (the output from an SCR dimmer, which consists of a sinewave which is at zero until part way through the wave).
I've been able to generate the waveform and perform FFT to get a set of Frequency vs Amplitude points, however I am not sure how to convert this data into bins suitable for generating a histogram.
Sample code and an image of what I'm after below - thanks for the help!
...ANSWER
Answered 2018-Jun-01 at 17:59You know your base frequency (fundamental tone), let's call it F
. 2*F
is the second harmonic, 3*F
the third, etc. You want to set histogram bin edges halfway between these: 1.5*F
, 2.5*F
, etc.
You have two periods in your input signal, therefore your (integer) base frequency is k=2
(the value at fft_vals[k+1]
, the first peak in your plot). The second harmonic is at k=4
, the third one at k=6
, etc.
So you would set your bins edges at k = 1:2:end
.
In general, this would be k = nCycles/2:nCycles:end
.
You can compute your bar graph according to our computed bin edges as follows:
QUESTION
I am creating 2 sinusoidal waves in python in order to test some algorithms and specially, measure and fix some delays between phases. It's supposed to be simulating mains voltage and current
...ANSWER
Answered 2018-Mar-05 at 16:01OK, I found the answer but forgot to reply here.
The most important thing is that the variable x is wrong, it's not supposed to be an np.arange(sample), this gives you an array from 0 to sample, which is just wrong. After I saw that I realized I was doing it wrong ... I simply did that because I use arange a lot and I ended up using it this way ... so I corrected to x = np.arange(0, 1/f, 1/Ts), where Ts = 1/fs. After I did that everything started working perfectly.
So ... a note for everyone, even if it's something you use/do daily, think before you just use/do it like you always do, some times you will prevent bugs and save yourself a few days of debugging
QUESTION
I'm trying to learn/build a 3-phase system monitoring device based on Arduino Mega, which will allow the user to monitor 220V/50Hz 3-phase system voltage and current with phase sequence detection for the correct wiring order (L1, l2, l3).
I was following this project from engineer experiences and achieved good results on the hardware part (schematics and circuit from link above).
For the software I used EmonLib for voltage and current measurements. I got the correct readings from Arduino, I think I don't understand the part related to adding timers loop to the code.
This is my modified code:
...ANSWER
Answered 2017-Dec-22 at 03:03int s1 = analogRead(A3);
int s2 = analogRead(A4);
int s3 = analogRead(A5);
QUESTION
Can anyone help me with my product cart page? I can't find where I'm making a mistake.
...ANSWER
Answered 2017-Jun-09 at 06:57Thanks for posting such great info to troubleshoot with, but without a little bit more it'll be hard for me to help. Have you thought of installing a chrome extension like Pesticide? It'll help you actually see outlines of what is where so you can figure out what happened. You can right click on elements and left click inspect to then see them in Chrome Developer Tools and actually look at what CSS is applying to them. You can then add in-line styles to style them manually while the window it open to try out solutions before returning to your CSS document and finalizing them (by adding them more permanently). Take caution when using !important though as it's kind of bringing a gun to a knife fight imho. Good luck and feel free to post screens of your site once pesticide is running and drawing boxes.
QUESTION
I have to take the FFT of a sin wave of 50 Hz and measure up to the 16 harmonics.
My sampling frequency is as per Nyquist criteria: fs = 16*50*2 = 1600 Hz = 1600 samples/sec
i.e in one period of 50Hz corresponds to 20 msec or 32 samples.
As shown in FIG-1, I will take 32 samples per cycle.
Then as per FIG-2 I will do the 32 point FFT of the ADC sample voltage series x[n]
where n
= 0 to 31
Then FFT algorithm returns the value X[k]
where k
= 0 to 31.
If
x[n]
is the sample of voltage.
So my question is in the output of the FFT algorithmX[k]
, where k = 0 to 31 :
X[0]
= fundamental frequency
X[1]
= 1st harmonic . . .
X[31]
= 31st harmonicIs it right ?
Also if value of
X[1]
= 1 + j, then magnitude ofX[1]
= sqrt(2) = 1.4142. So is this value 1.4142 the peak value of the first harmonic ? Now if I have to find RMS value of first harmonic then will it be Vrms = Vm/sqrt(2) = 1 ?Also should the 32 samples of the input signal start from zero crossing of the sin wave, or can I start at any time place of the sin wave as shown in figure-3?
- Also one more thing before feeding the 32 samples of the 50 Hz Signal to the FFT algorithm do I have to do some digital filtering on these samples .. why I am asking this question because suppose while taking 20 th & 16th sample if some noise spike comes then in that case it will not be the true value. If yes then which digital filtering method will be best ?
Please correct me.
...ANSWER
Answered 2017-May-25 at 12:52
X[0]
= fundamental frequency
X[1]
= 1st harmonic ...
X[31]
= 31st harmonicIs it right ?
Almost but not quite. You should not forget about the constant term (which shifts all your assumed correspondences up), and that any terms above the Nyquist frequency are mirrored from lower frequency components. The correspondence is as follows:
X[0]
is the constant term (also refereed to as DC bias)X[1]
is the fundamental frequencyX[2]
is the first harmonicX[16]
is the 15th harmonicX[17]
toX[31]
are the complex conjugates ofX[15]
toX[1]
respectively.
Also if value of
X[1] = 1 + j
, then magnitude ofX[1]
is sqrt(2) = 1.4142. So is this value 1.4142 the peak value of the first harmonic ?
Since your signal's frequency is an exact multiple of the FFT frequency bin width, the magnitude of X[1]
does correspond to the peak of the corresponding frequency component. As pointed out earlier X[1]
would correspond to the fundamental frequency, rather than the first harmonic. So the 1.4142 value in your example would be the peak value of the fundamental frequency.
Now if I have to find RMS value of first harmonic then will it be Vrms = Vm/sqrt(2) = 1 ?
That would indeed be the relationship between the RMS of a single harmonic and its peak value. However be careful not to apply this to convert between the RMS and the peak value of a signal with more than a single harmonic since the relationship is not linear.
Also should the 32 samples of the input signal start from zero crossing of the sin wave, or can I start at any time place of the sin wave as shown in figure-3?
You can start at any time offset in the periodic wave, but this will introduce phase offsets in the FFT results. If you are only interested in the magnitude of the frequency components, then it wouldn't matter.
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