I have the following LED array:
LED Circuit
I need help calculating the power supply needed. Rather than sticking it on an adjustable power supply, setting the current limit to 700mA and turning up the voltage I would prefer to learn how to calculate the optimal voltage. The LEDS and 1W and limited to 700mA and instead of just 5 columns my actual circuit has 14 columns. Please can someone teach me how to do the calculations for this circuit. If it were a straight forward parallel string of LEDs, I would not bother you, but the additional junctions throw me off a bit.
Please don't suggest alternative layouts as this is what I am reverse engineering.
Thanks,
Dylan
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I need to create a board that will contain about 50 LED’s. The LED’s need to be turned on/off individually, and together (timer is based on days suppose every Monday all the 50 LED’s turn on and every day of the week 10 LED’s are turned on).
In my research I have found the LM555 timer but that would lead to a huge circuit of 50 different timers
What is the most efficient way of controlling these LED’s
My first answer was really stupid (i'm so sorry about that) and i don't know why I didn't think before about the amazing "Shift Registers".
Your Arduino don't have so much pin to light every LEDs you had. But using shift registers this is possible. From an 8-bit shift register you can light 8 LEDs using just 2-3 PIN on your Arduino board (1 for clock, 1 for data and 1 for latch).
So to light 50 PINs you just need 7 of this components (for example).
How to use it? There are a lot of tutorials on internet, usually the sparkFun tutorial is my favorite.
Below: an image from Arduino site. It explained how to connect 16 LEDs.
Anyway from the software side, what you have to know ShiftOut function.
After that you have to use time function.
First of all initialize your time variables as you can see in link I posted above.
After, in your loop:
Put HIGH your Datapin connected to 50 leds when previousMonth != month.
Put HIGH dataPin connected to 10 leds when daySum == 10 (so, when previousDay != day you have to increase daySum using daySum++).
I have a 2 column vector with times and speeds of a subset of data, like so:
5 40
10 37
15 34
20 39
And so on. I want to get the fourier transform of speeds to get a frequency. How would I go about doing this with a fast fourier transform (fft)?
If my vector name is sampleData, I have tried
fft(sampleData);
but that gives me a vector of real and imaginary numbers. To be able to get sensible data to plot, how would I go about doing this?
Fourier Transform will yield a complex vector, when you fft you get a vector of frequencies, each has a spectral phase. These phases can be extremely important! (they contain most of the information of the time-domain signal, you won't see interference effects without them etc...). If you want to plot the power spectrum, you can
plot(abs(fft(sampleData)));
To complete the story, you'll probably need to fftshift, and also produce a frequency vector. Here's a more elaborate code:
% Assuming 'time' is the 1st col, and 'sampleData' is the 2nd col:
N=length(sampleData);
f=window(#hamming,N)';
dt=mean(diff(time));
df=1/(N*dt); % the frequency resolution (df=1/max_T)
if mod(N,2)==0
f_vec= df*((1:N)-1-N/2); % frequency vector for EVEN length vector
else
f_vec= df*((1:N)-0.5-N/2);
end
fft_data= fftshift(fft(fftshift(sampleData.*f))) ;
plot(f_vec,abs(fft_data))
I would recommend that you back up and think about what you are trying to accomplish, and whether an FFT is an appropriate tool for your situation. You say that you "want to ... get a frequency", but what exactly do you mean by that? Do you know that this data has exactly one frequency component, and want to know what the frequency is? Do you want to know both the frequency and phase of the component? Do you just want to get a rough idea of how many discrete frequency components are present? Are you interested in the spectrum of the noise in your measurement? There are many questions you can ask about "frequencies" in a data set, and whether or not an FFT and/or power spectrum is the best approach to getting an answer depends on the question.
In a comment above you asked "Is there some way to correlate the power spectrum to the time values?" This strikes me as a confused question, but also makes me think that maybe the question you are really trying to answer is "I have a signal whose frequency varies with time, and I want to get an estimate of the frequency vs time". I'm sure I've seen a question along those lines within the past few months here on SO, so I would search for that.
I would like to know how noise can be removed from data (say, radio data that is an array of rows and columns with each data point representing intensity of the radiation in the given frequency and time).The array can contain radio bursts. But many fixed frequency radio noise also exists(RFI=radio frequency intereference).How to remove such noise and bring out only the burst.
I don't mean to be rude, but this question isn't clear at all. Please sharpen it up.
The normal way to remove noise is first to define it exactly and then filter it out. Usually this is done in the frequency domain. For example, if you know the normalized power spectrum P(f) of the noise, build a filter with response
e/(e + P(f))
where e<1 is an attenuation factor.
You can implement the filter digitally using FFT or a convolution kernel.
When you don't know the spectrum of the noise or when it's white, then just use the inverse of the signal band.
I am making a finger plethysmograph(FP) using an LED and a receiver. The sensor produces an analog pulse waveform that is filtered, amplified and fed into a microcontroller input with a range of 3.3-0V. This signal is converted into its digital form.
Smapling rate is 8MHz, Processor frequency is 26MHz, Precision is 10 or 8 bit.
I am having problems coming up with a robust method for peak detection. I want to be able to detect heart pulses from the finger plethysmograph. I have managed to produce an accurate measurement of heart rate using a threshold method. However, the FP is extremely sensitive to movement and the offset of the signal can change based on movement. However, the peaks of the signal will still show up but with varying voltage offset.
Therefore, I am proposing a peak detection method that uses the slope to detect peaks. In example, if a peak is produced, the slope before and after the maximum point will be positive and negative respectively.
How feasible do you think this method is? Is there an easier way to perform peak detection using a microcontroller?
You can still introduce detection of false peaks when the device is moved. This will be present whether you are timing average peak duration or applying an FFT (fast Fourier Transform).
With an FFT you should be able to ignore peaks outside the range of frequencies you are considering (ie those < 30 bpm and > 300 bpm, say).
As Kenny suggests, 8MHz might overwhelm a 26MHz chip. Any particular reason for such a high sampling rate?
Like some of the comments, I would also recommend lowering your sample rate since you only care about pulse (i.e. heart rate) for now. So, assuming you're going to be looking at resting heart rate, you'll be in the sub-1Hz to 2Hz range (60 BPM = 1Hz), depending on subject health, age, etc.
In order to isolate the frequency range of interest, I would also recommend a simple, low-order digital filter. If you have access to Matlab, you can play around with Digital Filter Design using its Filter Design and Analysis Tool (Introduction to the FDATool). As you'll find out, Digital Filtering (wiki) is not computationally expensive since it is a matter of multiplication and addition.
To answer the detection part of your question, YES, it is certainly feasible to implement peak detection on the plethysmograph waveform within a microcontroller. Taking your example, a slope-based peak detection algorithm would operate on your waveform data, searching for changes in slope, essentially where the slope waveform crosses zero.
Here are a few other things to consider about your application:
Calculating slope can have a "spread" (i.e. do you find the slope between adjacent samples, or samples which are a few samples apart?)
What if your peak detection algorithm locates peaks that are too close together, or too far apart, in a physiological sense?
A Pulse Oximeter (wiki) often utilizes LEDs which emit Red and Infrared light. How does the frequency of the LED affect the plethysmograph? (HINT: It may not be significant, but I believe you'll find one wavelength to yield greater amplitudes in your frequency range of interest.)
Of course you'll find a variety of potential algorithms if you do a literature search but I think slope-based detection is great for its simplicity. Hope it helps.
If you can detect the period using zero crossing, even at 10x oversampling of 10 Hz, you can use a line fit of the quick-n-dirty-edge to find the exact period, and then subtract the new wave's samples in that period with the previous, and get a DC offset. The period measurement will have the precision of your sample rate. Doing operations on the time and amplitude-normalized data will be much easier.
This idea is computationally light compared to FFT, which still needs additional data processing.
Question: A PID controller has three parameters Kp, Ki and Kd which could affect the output performance. A differential driving robot is controlled by a PID controller. The heading information is sensed by a compass sensor. The moving forward speed is kept constant. The PID controller is able to control the heading information to follow a given direction. Explain the outcome on the differential driving robot performance when the three parameters are increased individually.
This is a question that has come up in a past paper but most likely won't show up this year but it still worries me. It's the only question that has me thinking for quite some time. I'd love an answer in simple terms. Most stuff i've read on the internet don't make much sense to me as it goes heavy into the detail and off topic for my case.
My take on this:
I know that the proportional term, Kp, is entirely based on the error and that, let's say, double the error would mean doubling Kp (applying proportional force). This therefore implies that increasing Kp is a result of the robot heading in the wrong direction so Kp is increased to ensure the robot goes on the right direction or at least tries to reduce the error as time passes so an increase in Kp would affect the robot in such a way to adjust the heading of the robot so it stays on the right path.
The derivative term, Kd, is based on the rate of change of the error so an increase in Kd implies that the rate of change of error has increased over time so double the error would result in double the force. An increase by double the change in the robot's heading would take place if the robot's heading is doubled in error from the previous feedback result. Kd causes the robot to react faster as the error increases.
An increase in the integral term, Ki, means that the error is increased over time. The integral accounts for the sum of error over time. Even a small increase in the error would increase the integral so the robot would have to head in the right direction for an equal amount of time for the integral to balance to zero.
I would appreciate a much better answer and it would be great to be confident for a similar upcoming question in the finals.
Side note: i've posted this question on the Robotics part earlier but seeing that the questions there are hardly ever noticed, i've also posted it here.
I would highly recommend reading this article PID Without a PhD it gives a great explanation along with some implementation details. The best part is the numerous graphs. They show you what changing the P, I, or D term does while holding the others constant.
And if you want real world Application Atmel provides example code on their site (for 8 bit MCU) that perfectly mirrors the PID without a PhD article. It follows this code from AVR's website exactly (they make the ATMega32p microcontroller chip on the Arduino UNO boards) PDF explanation and Atmel Code in C
But here is a general explanation the way I understand it.
Proportional: This is a proportional relationship between the error and the target. Something like Pk(target - actual) Its simply a scaling factor on the error. It decides how quickly the system should react to an error (if it is of any help, you can think of it like amplifier slew rate). A large value will quickly try to fix errors, and a slow value will take longer. With Higher values though, we get into an overshoot condition and that's where the next terms come into play
Integral: This is meant to account for errors in the past. In fact it is the sum of all past errors. This is often useful for things like a small dc/constant offset that a Proportional controller can't fix on its own. Imagine, you give a step input of 1, and after a while the output settles at .9 and its clear its not going anywhere. The integral portion will see this error is always ~.1 too small so it will add it back in, to hopefully bring control closer to 1. THis term usually helps to stabilize the response curve. Since it is taken over a long period of time, it should reduce noise and any fast changes (like those found in overshoot/ringing conditions). Because it's aggregate, it is a very sensitive measurement and is usually very small when compared to other terms. A lower value will make changes happen very slowly, and create a very smooth response(this can also cause "wind-up" see the article)
Derivative: This is supposed to account for the "future". It uses the slope of the most recent samples. Remember this is the slope, it has nothing to do with the position error(current-goal), it is previous measured position - current measured position. This is most sensitive to noise and when it is too high often causes ringing. A higher value encourages change since we are "amplifying" the slope.
I hope that helps. Maybe someone else can offer another viewpoint, but that's typically how I think about it.