Moving average digital filter implementation - filter

I need to implement moving average digital filter for post processing of some
recorded oscilloscope waveforms in Scilab. I have prepared a script with
below given code (the recursive implementation with averaging window containing 256 samples)
// number of samples
N = 350000;
// vector of voltage samples
voltage = M(1:N, 2)';
// filtered values
filt_voltage = zeros(1:N);
// window length
L = 256;
// sum of the samples in the averaging window
sum = 0
for i = 1:N_01
// averaging window full?
if i > L
// remove the oldest sample in the averaging window
sum = sum - voltage(i - L);
end
// add the newest sample into the averaging window
sum = sum + voltage(i);
// average of the samples in the averaging window
filt_voltage(i) = sum/L;
end
The script output is following (blue waveform - the recorded data, red waveform - filtered data)
The problem is that I am not sure whether me moving average implementation
is correct (I have found a lot of implementations mostly based on convolution). The output seems to be somehow filtered but it would be helpful
for me if anybody can confirm to me that it is correct. Thanks in advance.

There is nothing wrong with your implementation. In fact, it is a common implementation of the recursive formulation given on stats.stackexchange.com and on wikipedia:

Related

How to calculate the length of cable on a winch given the rotations of the drum

I have a cable winch system that I would like to know how much cable is left given the number of rotations that have occurred and vice versa. This system will run on a low-cost microcontroller with low computational resources and should be able to update quickly, long for/while loop iterations are not ideal.
The inputs are cable diameter, inner drum diameter, inner drum width, and drum rotations. The output should be the length of the cable on the drum.
At first, I was calculating the maximum number of wraps of cable per layer based on cable diameter and inner drum width, I could then use this to calculate the length of cable per layer. The issue comes when I calculate the total length as I have to loop through each layer, a costly operation (could be 100's of layers).
My next approach was to precalculate a table with each layer, then perform a 3-5 degree polynomial regression down to an easy to calculate formula.
This appears to work for the most part, however, there are slight inaccuracies at the low and high end (0 rotations could be + or - a few units of cable length). The real issue comes when I try and reverse the function to get the current rotations of the drum given the length. So far, my reversed formula does not seem to equal the forward formula (I am reversing X and Y before calculating the polynomial).
I have looked high and low and cannot seem to find any formulas for cable length to rotations that do not use recursion or loops. I can't figure out how to reverse my polynomial function to get the reverse value without losing precision. If anyone happens to have an insight/ideas or can help guide me in the right direction that would be most helpful. Please see my attempts below.
// Units are not important
CableLength = 15000
CableDiameter = 5
DrumWidth = 50
DrumDiameter = 5
CurrentRotations = 0
CurrentLength = 0
CurrentLayer = 0
PolyRotations = Array
PolyLengths = Array
PolyLayers = Array
WrapsPerLayer = DrumWidth / CableDiameter
While CurrentLength < CableLength // Calcuate layer length for each layer up to cable length
CableStackHeight = CableDiameter * CurrentLayer
DrumDiameterAtLayer = DrumDiameter + (CableStackHeight * 2) // Assumes cables stack vertically
WrapDiameter = DrumDiameterAtLayer + CableDiameter // Center point of cable
WrapLength = WrapDiameter * PI
LayerLength = WrapLength * WrapsPerLayer
CurrentRotations += WrapsPerLayer // 1 Rotation per wrap
CurrentLength += LayerLength
CurrentLayer++
PolyRotations.Push(CurrentRotations)
PolyLengths.Push(CurrentLength)
PolyLayers.Push(CurrentLayer)
End
// Using 5 degree polynomials, any lower = very low precision
PolyLengthToRotation = CreatePolynomial(PolyLengths, PolyRotations, 5) // 5 Degrees
PolyRotationToLength = CreatePolynomial(PolyRotations, PolyLengths, 5) // 5 Degrees
// 40 Rotations should equal about 3141.593 units
RealRotation = 40
RealLength = 3141.593
CalculatedLength = EvaluatePolynomial(RealRotation,PolyRotationToLength)
CalculatedRotations = EvaluatePolynomial(RealLength,PolyLengthToRotation)
// CalculatedLength = 3141.593 // Good
// CalculatedRotations = 41.069 // No good
// CalculatedRotations != RealRotation // These should equal
// 0 Rotations should equal 0 length
RealRotation = 0
RealLength = 0
CalculatedLength = EvaluatePolynomial(RealRotation,PolyRotationToLength)
CalculatedRotations = EvaluatePolynomial(RealLength,PolyLengthToRotation)
// CalculatedLength = 1.172421e-9 // Very close
// CalculatedRotations = 1.947, // No good
// CalculatedRotations != RealRotation // These should equal
Side note: I have a "spool factor" parameter to calibrate for the actual cable spooling efficiency that is not shown here. (cable is not guaranteed to lay mathematically perfect)
#Bathsheba May have meant cable, but a table is a valid option (also experimental numbers are probably more interesting in the real world).
A bit slow, but you could always do it manually. There's only 40 rotations (though optionally for better experimental results, repeat 3 times and take the average...). Reel it completely in. Then do a rotation (depending on the diameter of your drum, half rotation). Measure and mark how far it spooled out (tape), record it. Repeat for the next 39 rotations. You now have a lookup table you can find the length in O(log N) via binary search (by sorting the data) and a bit of interpolation (IE: 1.5 rotations is about half way between 1 and 2 rotations).
You can also use this to derived your own experimental data. Do the same thing, but with a cable half as thin (perhaps proportional to the ratio of the inner diameter and the cable radius?). What effect does it have on the numbers? How about twice or half the diameter? Math says circumference is linear (2πr), so half the radius, half the amount per rotation. Might be easier to adjust the table data.
The gist is that it may be easier for you to have a real world reference for your numbers rather than relying purely on an abstract mathematically model (not to say the model would be wrong, but cables don't exactly always wind up perfectly, who knows perhaps you can find a quirk about your winch that would have lead to errors in a pure mathematical approach). Who knows might be able to derive the formula yourself :) with a fudge factor for the real world even lol.

Real-time peak detection in noisy sinusoidal time-series

I have been attempting to detect peaks in sinusoidal time-series data in real time, however I've had no success thus far. I cannot seem to find a real-time algorithm that works to detect peaks in sinusoidal signals with a reasonable level of accuracy. I either get no peaks detected, or I get a zillion points along the sine wave being detected as peaks.
What is a good real-time algorithm for input signals that resemble a sine wave, and may contain some random noise?
As a simple test case, consider a stationary, sine wave that is always the same frequency and amplitude. (The exact frequency and amplitude don't matter; I have arbitrarily chosen a frequency of 60 Hz, an amplitude of +/− 1 unit, at a sampling rate of 8 KS/s.) The following MATLAB code will generate such a sinusoidal signal:
dt = 1/8000;
t = (0:dt:(1-dt)/4)';
x = sin(2*pi*60*t);
Using the algorithm developed and published by Jean-Paul, I either get no peaks detected (left) or a zillion "peaks" detected (right):
I've tried just about every combination of values for these 3 parameters that I could think of, following the "rules of thumb" that Jean-Paul gives, but I have so far been unable to get my expected result.
I found an alternative algorithm, developed and published by Eli Billauer, that does give me the results that I want—e.g.:
Even though Eli Billauer's algorithm is much simpler and does tend to reliably produce the results that I want, it is not suitable for real-time applications.
As another example of a signal that I'd like to apply such an algorithm to, consider the test case given by Eli Billauer for his own algorithm:
t = 0:0.001:10;
x = 0.3*sin(t) + sin(1.3*t) + 0.9*sin(4.2*t) + 0.02*randn(1, 10001);
This is a more unusual (less uniform/regular) signal, with a varying frequency and amplitude, but still generally sinusoidal. The peaks are plainly obvious to the eye when plotted, but hard to identify with an algorithm.
What is a good real-time algorithm to correctly identify the peaks in a sinusoidal input signal? I am not really an expert when it comes to signal processing, so it would be helpful to get some rules of thumb that consider sinusoidal inputs. Or, perhaps I need to modify e.g. Jean-Paul's algorithm itself in order to work properly on sinusoidal signals. If that's the case, what modifications would be required, and how would I go about making these?
Case 1: sinusoid without noise
If your sinusoid does not contain any noise, you can use a very classic signal processing technique: taking the first derivative and detecting when it is equal to zero.
For example:
function signal = derivesignal( d )
% Identify signal
signal = zeros(size(d));
for i=2:length(d)
if d(i-1) > 0 && d(i) <= 0
signal(i) = +1; % peak detected
elseif d(i-1) < 0 && d(i) >= 0
signal(i) = -1; % trough detected
end
end
end
Using your example data:
% Generate data
dt = 1/8000;
t = (0:dt:(1-dt)/4)';
y = sin(2*pi*60*t);
% Add some trends
y(1:1000) = y(1:1000) + 0.001*(1:1000)';
y(1001:2000) = y(1001:2000) - 0.002*(1:1000)';
% Approximate first derivative (delta y / delta x)
d = [0; diff(y)];
% Identify signal
signal = derivesignal(d);
% Plot result
figure(1); clf; set(gcf,'Position',[0 0 677 600])
subplot(4,1,1); hold on;
title('Data');
plot(t,y);
subplot(4,1,2); hold on;
title('First derivative');
area(d);
ylim([-0.05, 0.05]);
subplot(4,1,3); hold on;
title('Signal (-1 for trough, +1 for peak)');
plot(t,signal); ylim([-1.5 1.5]);
subplot(4,1,4); hold on;
title('Signals marked on data');
markers = abs(signal) > 0;
plot(t,y); scatter(t(markers),y(markers),30,'or','MarkerFaceColor','red');
This yields:
This method will work extremely well for any type of sinusoid, with the only requirement that the input signal contains no noise.
Case 2: sinusoid with noise
As soon as your input signal contains noise, the derivative method will fail. For example:
% Generate data
dt = 1/8000;
t = (0:dt:(1-dt)/4)';
y = sin(2*pi*60*t);
% Add some trends
y(1:1000) = y(1:1000) + 0.001*(1:1000)';
y(1001:2000) = y(1001:2000) - 0.002*(1:1000)';
% Add some noise
y = y + 0.2.*randn(2000,1);
Will now generate this result because first differences amplify noise:
Now there are many ways to deal with noise, and the most standard way is to apply a moving average filter. One disadvantage of moving averages is that they are slow to adapt to new information, such that signals may be identified after they have occurred (moving averages have a lag).
Another very typical approach is to use Fourier Analysis to identify all the frequencies in your input data, disregard all low-amplitude and high-frequency sinusoids, and use the remaining sinusoid as a filter. The remaining sinusoid will be (largely) cleansed from the noise and you can then use first-differencing again to determine the peaks and troughs (or for a single sine wave you know the peaks and troughs happen at 1/4 and 3/4 pi of the phase). I suggest you pick up any signal processing theory book to learn more about this technique. Matlab also has some educational material about this.
If you want to use this algorithm in hardware, I would suggest you also take a look at WFLC (Weighted Fourier Linear Combiner) with e.g. 1 oscillator or PLL (Phase-Locked Loop) that can estimate the phase of a noisy wave without doing a full Fast Fourier Transform. You can find a Matlab algorithm for a phase-locked loop on Wikipedia.
I will suggest a slightly more sophisticated approach here that will identify the peaks and troughs in real-time: fitting a sine wave function to your data using moving least squares minimization with initial estimates from Fourier analysis.
Here is my function to do that:
function [result, peaks, troughs] = fitsine(y, t, eps)
% Fast fourier-transform
f = fft(y);
l = length(y);
p2 = abs(f/l);
p1 = p2(1:ceil(l/2+1));
p1(2:end-1) = 2*p1(2:end-1);
freq = (1/mean(diff(t)))*(0:ceil(l/2))/l;
% Find maximum amplitude and frequency
maxPeak = p1 == max(p1(2:end)); % disregard 0 frequency!
maxAmplitude = p1(maxPeak); % find maximum amplitude
maxFrequency = freq(maxPeak); % find maximum frequency
% Initialize guesses
p = [];
p(1) = mean(y); % vertical shift
p(2) = maxAmplitude; % amplitude estimate
p(3) = maxFrequency; % phase estimate
p(4) = 0; % phase shift (no guess)
p(5) = 0; % trend (no guess)
% Create model
f = #(p) p(1) + p(2)*sin( p(3)*2*pi*t+p(4) ) + p(5)*t;
ferror = #(p) sum((f(p) - y).^2);
% Nonlinear least squares
% If you have the Optimization toolbox, use [lsqcurvefit] instead!
options = optimset('MaxFunEvals',50000,'MaxIter',50000,'TolFun',1e-25);
[param,fval,exitflag,output] = fminsearch(ferror,p,options);
% Calculate result
result = f(param);
% Find peaks
peaks = abs(sin(param(3)*2*pi*t+param(4)) - 1) < eps;
% Find troughs
troughs = abs(sin(param(3)*2*pi*t+param(4)) + 1) < eps;
end
As you can see, I first perform a Fourier transform to find initial estimates of the amplitude and frequency of the data. I then fit a sinusoid to the data using the model a + b sin(ct + d) + et. The fitted values represent a sine wave of which I know that +1 and -1 are the peaks and troughs, respectively. I can therefore identify these values as the signals.
This works very well for sinusoids with (slowly changing) trends and general (white) noise:
% Generate data
dt = 1/8000;
t = (0:dt:(1-dt)/4)';
y = sin(2*pi*60*t);
% Add some trends
y(1:1000) = y(1:1000) + 0.001*(1:1000)';
y(1001:2000) = y(1001:2000) - 0.002*(1:1000)';
% Add some noise
y = y + 0.2.*randn(2000,1);
% Loop through data (moving window) and fit sine wave
window = 250; % How many data points to consider
interval = 10; % How often to estimate
result = nan(size(y));
signal = zeros(size(y));
for i = window+1:interval:length(y)
data = y(i-window:i); % Get data window
period = t(i-window:i); % Get time window
[output, peaks, troughs] = fitsine(data,period,0.01);
result(i-interval:i) = output(end-interval:end);
signal(i-interval:i) = peaks(end-interval:end) - troughs(end-interval:end);
end
% Plot result
figure(1); clf; set(gcf,'Position',[0 0 677 600])
subplot(4,1,1); hold on;
title('Data');
plot(t,y); xlim([0 max(t)]); ylim([-4 4]);
subplot(4,1,2); hold on;
title('Model fit');
plot(t,result,'-k'); xlim([0 max(t)]); ylim([-4 4]);
subplot(4,1,3); hold on;
title('Signal (-1 for trough, +1 for peak)');
plot(t,signal,'r','LineWidth',2); ylim([-1.5 1.5]);
subplot(4,1,4); hold on;
title('Signals marked on data');
markers = abs(signal) > 0;
plot(t,y,'-','Color',[0.1 0.1 0.1]);
scatter(t(markers),result(markers),30,'or','MarkerFaceColor','red');
xlim([0 max(t)]); ylim([-4 4]);
Main advantages of this approach are:
You have an actual model of your data, so you can predict signals in the future before they happen! (e.g. fix the model and calculate the result by inputting future time periods)
You don't need to estimate the model every period (see parameter interval in the code)
The disadvantage is that you need to select a lookback window, but you will have this problem with any method that you use for real-time detection.
Video demonstration
Data is the input data, Model fit is the fitted sine wave to the data (see code), Signal indicates the peaks and troughs and Signals marked on data gives an impression of how accurate the algorithm is. Note: watch the model fit adjust itself to the trend in the middle of the graph!
That should get you started. There are also a lot of excellent books on signal detection theory (just google that term), which will go much further into these types of techniques. Good luck!
Consider using findpeaks, it is fast, which may be important for realtime. You should filter high-frequency noise to improve accuracy. here I smooth the data with a moving window.
t = 0:0.001:10;
x = 0.3*sin(t) + sin(1.3*t) + 0.9*sin(4.2*t) + 0.02*randn(1, 10001);
[~,iPeak0] = findpeaks(movmean(x,100),'MinPeakProminence',0.5);
You can time the process (0.0015sec)
f0 = #() findpeaks(movmean(x,100),'MinPeakProminence',0.5)
disp(timeit(f0,2))
To compare, processing the slope is only a bit faster (0.00013sec), but findpeaks have many useful options, such as minimum interval between peaks etc.
iPeaks1 = derivePeaks(x);
f1 = #() derivePeaks(x)
disp(timeit(f1,1))
Where derivePeaks is:
function iPeak1 = derivePeaks(x)
xSmooth = movmean(x,100);
goingUp = find(diff(movmean(xSmooth,100)) > 0);
iPeak1 = unique(goingUp([1,find(diff(goingUp) > 100),end]));
iPeak1(iPeak1 == 1 | iPeak1 == length(iPeak1)) = [];
end

Detect Greatest Dynamic Range Changes In Audio Sample

Forgive/correct any wrong terminology in the below (I hope it makes sense!):
I want to detect the biggest dynamic audio changes in a given sample, (ie. the moments when the sound wave 'grows'/'accelerates' the most).
For example, if the audio goes quiet at some points during the sample, I want to know when the music comes back in after, and order these data points by the relative dynamic range (volume?) increase (largest to smallest).
My audio sample is a buffer of float32[] and sample rate, and I would like a resulting array of objects each containing:
start frame index
start time (seconds ... frameIndex/sampleRate?)
end frame index
end time (seconds)
dynamic change value
My naive approach iterates linearly and detects points at which the value starts rising until it is no longer rising, and then calculates the rise over run for each sub interval between those points.. but this is not producing the correct result.
Any ideas or existing algorithms that do this?
Not picky on languages, but anything with syntax like C#, Java, JavaScript preferred!
I am a little unsure as to how much audio DSP background you have so apologies if treading over old territory.
Essentially this is a problem of trying to find the envelope of the signal at any given point.
Since the audio signal will be fluctuating between -1 and 1, the value of any individual sample will not yield much
information about the loudness or dynamic range.
What would be best to find is the root mean square of the signal over some frame of audio data
Written in pseudo code, and assuming you already have your audio Data, a function and way of grabbing the rms data could be:
function rms(frame[], frameSize)
{
var rmsValue = 0;
for(int i = 0; i < frameSize; i++)
{
rmsValue += frame[i] * frame[i]; // square the sample and sum over frame
}
rmsValue = sqrt(rmsValue / frameSize);
return rmsValue;
}
// Main
var frameNum = floor(numberOfAudioSample / frameSize) // for analysis just floor to a whole number of frames, if thi is real-time, you will need to deal with a partial frame at the end
var frame = [] // an array or buffer to temporarily store audio data
var rmsData = [] // an array or buffer to store RMS data
for (var i = 0; i < frameNum; i++)
{
for (var j = 0; j < frameSize; j++)
{
sampleIndex = j + (i * frameSize)
frame[j] = audioData[sampleIndex]
}
rmsData[i] = rms(frame, frameSize)
}
You can then compare elements of the RMS Data to find when the dynamics are changing and by how much.
For digital audio RMS will be constrained to between 0 and 1. To get dBFS then all you need to do is 20 * log10(rmsData)
Finding the exact sample where dynamic range changes will be tricky. The frame index should be accurate enough with a small enough size of frame.
The smaller the frame, however, the more erratic the RMS values will be. Finding a time in seconds is simply sampleIndex / samplingRate
With a small frame size you may also want to low pass filter the rms data. It depends on whether this is for a real-time application or for non real-time analysis.
To make things easy, I would prototype something in Octave or MATLAB first

Average set of color images and standard deviation

I am learning image analysis and trying to average set of color images and get standard deviation at each pixel
I have done this, but it is not by averaging RGB channels. (for ex rchannel = I(:,:,1))
filelist = dir('dir1/*.jpg');
ims = zeros(215, 300, 3);
for i=1:length(filelist)
imname = ['dir1/' filelist(i).name];
rgbim = im2double(imread(imname));
ims = ims + rgbim;
end
avgset1 = ims/length(filelist);
figure;
imshow(avgset1);
I am not sure if this is correct. I am confused as to how averaging images is useful.
Also, I couldn't get the matrix holding standard deviation.
Any help is appreciated.
If you are concerned about finding the mean RGB image, then your code is correct. What I like is that you converted the images using im2double before accumulating the mean and so you are making everything double precision. As what Parag said, finding the mean image is very useful especially in machine learning. It is common to find the mean image of a set of images before doing image classification as it allows the dynamic range of each pixel to be within a normalized range. This allows the training of the learning algorithm to converge quickly to the optimum solution and provide the best set of parameters to facilitate the best accuracy in classification.
If you want to find the mean RGB colour which is the average colour over all images, then no your code is not correct.
You have summed over all channels individually which is stored in sumrgbims, so the last step you need to do now take this image and sum over each channel individually. Two calls to sum in the first and second dimensions chained together will help. This will produce a 1 x 1 x 3 vector, so using squeeze after this to remove the singleton dimensions and get a 3 x 1 vector representing the mean RGB colour over all images is what you get.
Therefore:
mean_colour = squeeze(sum(sum(sumrgbims, 1), 2));
To address your second question, I'm assuming you want to find the standard deviation of each pixel value over all images. What you will have to do is accumulate the square of each image in addition to accumulating each image inside the loop. After that, you know that the standard deviation is the square root of the variance, and the variance is equal to the average sum of squares subtracted by the mean squared. We have the mean image, now you just have to square the mean image and subtract this with the average sum of squares. Just to be sure our math is right, supposing we have a signal X with a mean mu. Given that we have N values in our signal, the variance is thus equal to:
Source: Science Buddies
The standard deviation would simply be the square root of the above result. We would thus calculate this for each pixel independently. Therefore you can modify your loop to do that for you:
filelist = dir('set1/*.jpg');
sumrgbims = zeros(215, 300, 3);
sum2rgbims = sumrgbims; % New - for standard deviation
for i=1:length(filelist)
imname = ['set1/' filelist(i).name];
rgbim = im2double(imread(imname));
sumrgbims = sumrgbims + rgbim;
sum2rgbims = sum2rgbims + rgbim.^2; % New
end
rgbavgset1 = sumrgbims/length(filelist);
% New - find standard deviation
rgbstdset1 = ((sum2rgbims / length(filelist)) - rgbavgset.^2).^(0.5);
figure;
imshow(rgbavgset1, []);
% New - display standard deviation image
figure;
imshow(rgbstdset1, []);
Also to make sure, I've scaled the display of each imshow call so the smallest value gets mapped to 0 and the largest value gets mapped to 1. This does not change the actual contents of the images. This is just for display purposes.

How to compute frequency of data using FFT?

I want to know the frequency of data. I had a little bit idea that it can be done using FFT, but I am not sure how to do it. Once I passed the entire data to FFT, then it is giving me 2 peaks, but how can I get the frequency?
Thanks a lot in advance.
Here's what you're probably looking for:
When you talk about computing the frequency of a signal, you probably aren't so interested in the component sine waves. This is what the FFT gives you. For example, if you sum sin(2*pi*10x)+sin(2*pi*15x)+sin(2*pi*20x)+sin(2*pi*25x), you probably want to detect the "frequency" as 5 (take a look at the graph of this function). However, the FFT of this signal will detect the magnitude of 0 for the frequency 5.
What you are probably more interested in is the periodicity of the signal. That is, the interval at which the signal becomes most like itself. So most likely what you want is the autocorrelation. Look it up. This will essentially give you a measure of how self-similar the signal is to itself after being shifted over by a certain amount. So if you find a peak in the autocorrelation, that would indicate that the signal matches up well with itself when shifted over that amount. There's a lot of cool math behind it, look it up if you are interested, but if you just want it to work, just do this:
Window the signal, using a smooth window (a cosine will do. The window should be at least twice as large as the largest period you want to detect. 3 times as large will give better results). (see http://zone.ni.com/devzone/cda/tut/p/id/4844 if you are confused).
Take the FFT (however, make sure the FFT size is twice as big as the window, with the second half being padded with zeroes. If the FFT size is only the size of the window, you will effectively be taking the circular autocorrelation, which is not what you want. see https://en.wikipedia.org/wiki/Discrete_Fourier_transform#Circular_convolution_theorem_and_cross-correlation_theorem )
Replace all coefficients of the FFT with their square value (real^2+imag^2). This is effectively taking the autocorrelation.
Take the iFFT
Find the largest peak in the iFFT. This is the strongest periodicity of the waveform. You can actually be a little more clever in which peak you pick, but for most purposes this should be enough. To find the frequency, you just take f=1/T.
Suppose x[n] = cos(2*pi*f0*n/fs) where f0 is the frequency of your sinusoid in Hertz, n=0:N-1, and fs is the sampling rate of x in samples per second.
Let X = fft(x). Both x and X have length N. Suppose X has two peaks at n0 and N-n0.
Then the sinusoid frequency is f0 = fs*n0/N Hertz.
Example: fs = 8000 samples per second, N = 16000 samples. Therefore, x lasts two seconds long.
Suppose X = fft(x) has peaks at 2000 and 14000 (=16000-2000). Therefore, f0 = 8000*2000/16000 = 1000 Hz.
If you have a signal with one frequency (for instance:
y = sin(2 pi f t)
With:
y time signal
f the central frequency
t time
Then you'll get two peaks, one at a frequency corresponding to f, and one at a frequency corresponding to -f.
So, to get to a frequency, can discard the negative frequency part. It is located after the positive frequency part. Furthermore, the first element in the array is a dc-offset, so the frequency is 0. (Beware that this offset is usually much more than 0, so the other frequency components might get dwarved by it.)
In code: (I've written it in python, but it should be equally simple in c#):
import numpy as np
from pylab import *
x = np.random.rand(100) # create 100 random numbers of which we want the fourier transform
x = x - mean(x) # make sure the average is zero, so we don't get a huge DC offset.
dt = 0.1 #[s] 1/the sampling rate
fftx = np.fft.fft(x) # the frequency transformed part
# now discard anything that we do not need..
fftx = fftx[range(int(len(fftx)/2))]
# now create the frequency axis: it runs from 0 to the sampling rate /2
freq_fftx = np.linspace(0,2/dt,len(fftx))
# and plot a power spectrum
plot(freq_fftx,abs(fftx)**2)
show()
Now the frequency is located at the largest peak.
If you are looking at the magnitude results from an FFT of the type most common used, then a strong sinusoidal frequency component of real data will show up in two places, once in the bottom half, plus its complex conjugate mirror image in the top half. Those two peaks both represent the same spectral peak and same frequency (for strictly real data). If the FFT result bin numbers start at 0 (zero), then the frequency of the sinusoidal component represented by the bin in the bottom half of the FFT result is most likely.
Frequency_of_Peak = Data_Sample_Rate * Bin_number_of_Peak / Length_of_FFT ;
Make sure to work out your proper units within the above equation (to get units of cycles per second, per fortnight, per kiloparsec, etc.)
Note that unless the wavelength of the data is an exact integer submultiple of the FFT length, the actual peak will be between bins, thus distributing energy among multiple nearby FFT result bins. So you may have to interpolate to better estimate the frequency peak. Common interpolation methods to find a more precise frequency estimate are 3-point parabolic and Sinc convolution (which is nearly the same as using a zero-padded longer FFT).
Assuming you use a discrete Fourier transform to look at frequencies, then you have to be careful about how to interpret the normalized frequencies back into physical ones (i.e. Hz).
According to the FFTW tutorial on how to calculate the power spectrum of a signal:
#include <rfftw.h>
...
{
fftw_real in[N], out[N], power_spectrum[N/2+1];
rfftw_plan p;
int k;
...
p = rfftw_create_plan(N, FFTW_REAL_TO_COMPLEX, FFTW_ESTIMATE);
...
rfftw_one(p, in, out);
power_spectrum[0] = out[0]*out[0]; /* DC component */
for (k = 1; k < (N+1)/2; ++k) /* (k < N/2 rounded up) */
power_spectrum[k] = out[k]*out[k] + out[N-k]*out[N-k];
if (N % 2 == 0) /* N is even */
power_spectrum[N/2] = out[N/2]*out[N/2]; /* Nyquist freq. */
...
rfftw_destroy_plan(p);
}
Note it handles data lengths that are not even. Note particularly if the data length is given, FFTW will give you a "bin" corresponding to the Nyquist frequency (sample rate divided by 2). Otherwise, you don't get it (i.e. the last bin is just below Nyquist).
A MATLAB example is similar, but they are choosing the length of 1000 (an even number) for the example:
N = length(x);
xdft = fft(x);
xdft = xdft(1:N/2+1);
psdx = (1/(Fs*N)).*abs(xdft).^2;
psdx(2:end-1) = 2*psdx(2:end-1);
freq = 0:Fs/length(x):Fs/2;
In general, it can be implementation (of the DFT) dependent. You should create a test pure sine wave at a known frequency and then make sure the calculation gives the same number.
Frequency = speed/wavelength.
Wavelength is the distance between the two peaks.

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