What is a simple way to see if my low-pass filter is working? I'm in the process of designing a low-pass filter and would like to run tests on it in a relatively straight forward manner.
Presently I open up a WAV file and stick all the samples in a array of ints. I then run the array through the low-pass filter to create a new array. What would an easy way to check if the low-pass filter worked?
All of this is done in C.
You can use a broadband signal such as white noise to measure the frequency response:
generate white noise input signal
pass white noise signal through filter
take FFT of output from filter
compute log magnitude of FFT
plot log magnitude
Rather than coding this all up you can just dump the output from the filter to a text file and then do the analysis in e.g. MATLAB or Octave (hint: use periodogram).
Depends on what you want to test. I'm not a DSP expert, but I know there are different things one could measure about your filter (if that's what you mean by testing).
If the filter is linear then all information of the filter can be found in the impulse response. Read about it here: http://en.wikipedia.org/wiki/Linear_filter
E.g. if you take the Fourier transform of the impulse response, you'll get the frequency response. The frequency response easily tells you if the low-pass filter is worth it's name.
Maybe I underestimate your knowledge about DSP, but I recommend you to read the book on this website: http://www.dspguide.com. It's a very accessible book without difficult math. It's available as a real book, but you can also read it online for free.
EDIT: After reading it I'm convinced that every programmer that ever touches an ADC should definitely have read this book first. I discovered that I did a lot of things the difficult way in past projects that I could have done a thousand times better when I had a little bit more knowledge about DSP. Most of the times an unexperienced programmer is doing DSP without knowing it.
Create two monotone signals, one of a low frequency and one of a high frequency. Then run your filter on the two. If it works, then the low frequency signal should be unmodified whereas the high frequency signal will be filtered out.
Like Bart above mentioned.
If it's LTI system, I would insert impulse and record the samples and perform FFT using matlab and plot magnitude.
You ask why?
In time domain, you have to convolute the input x(t) with the impulse response d(t) to get the transfer function which is tedious.
y(t) = x(t) * d(t)
In frequency domain, convolution becomes simple multiplication.
y(s) = x(s) x d(s)
So, transfer function is y(s)/x(s) = d(s).
That's the reason you take FFT of impulse response to see the behavior of the filter.
You should be able to programmatically generate tones (sine waves) of various frequencies, stuff them into the input array, and then compare the signal energy by summing the squared values of the arrays (and dividing by the length, though that's mathematically not necessary here because the signals should be the same length). The ratio of the output energy to the input energy gives you the filter gain. If your LPF is working correctly, the gain should be close to 1 for low frequencies, close to 0.5 at the bandwidth frequency, and close to zero for high frequencies.
A note: There are various (but essentially the same in spirit) definitions of "bandwidth" and "gain". The method I've suggested should be relatively insensitive to the transient response of the filter because it's essentially averaging the intensity of the signal, though you could improve it by ignoring the first T samples of the input, where T is related to the filter bandwidth. Either way, make sure that the signals are long compared to the inverse of the filter bandwidth.
When I check a digital filter, I compute the magnitude response graph for the filter and plot it. I then generate a linear sweeping sine wave in code or using Audacity, and pass the sweeping sine wave through the filter (taking into account that things might get louder, so the sine wave is quiet enough not to clip) . A visual check is usually enough to assert that the filter is doing what I think it should. If you don't know how to compute the magnitude response I suspect there are tools out there that will compute it for you.
Depending on how certain you want to be, you don't even have to do that. You can just process the linear sweep and see that it attenuated the higher frequencies.
Related
Kalman filters and quaternions are something new for me.
I have a sensor which output voltage on its pins changes in function of its inclination on x,y and/or z-axis, i.e. an angle sensor.
My questions:
Is it possible to apply a Kalman filter to smooth the results and avoid any noise on the measurements?
I will then only have 1 single 3D vector. What kind of operations with quaternions could I use with this 3d vector to learn more about quaternions?
You can apply a Kalman filter to accelerometer data, it's a powerful technique though and there are lots of ways to do it wrong. If your goal is to learn about the filter then go for it - the discussion here might be helpful.
If you just want to smooth the data and get on with the next problem then you might want to start with a moving average filter, or traditional lowpass/bandpass filters.
After applying a Kalman filter you will still have a sequence of data - it won't reduce it to a single vector. If this is your goal you might as well take the mean of each coordinate sequence.
As for quaternions, you could probably come up with a way of performing quaternion operations on your accelerometer data but the challenge would be to make it meaningful. For the purposes of learning about the concept you really need it to make some sense, so that you can visualise the results and interpret them.
I would be tempted to write some functions to implement quaternion operations instead - multiplication is the strange one. This will be a good introduction to the way they work, and then when you find an application that calls for them you can use your functions and you'll already know how the mechanics work.
If you want to read the most famous use of quaternions have a look at Maxwell's equations in their original quaternion form, before Heaviside dramatically simplified them and put them in the vector notation we use today.
Also a lot of work is done using tensors these days and if you're interested in the more complex mathematical datatypes that would be a worthwhile one to look into.
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.
Given is an array of 320 elements (int16), which represent an audio signal (16-bit LPCM) of 20 ms duration. I am looking for a most simple and very fast method which should decide whether this array contains active audio (like speech or music), but not noise or silence. I don't need a very high quality of the decision, but it must be very fast.
It occurred to me first to add all squares or absolute values of the elements and compare their sum with a threshold, but such a method is very slow on my system, even if it is O(n).
You're not going to get much faster than a sum-of-squares approach.
One optimization that you may not be doing so far is to use a running total. That is, in each time step, instead of summing the squares of the last n samples, keep a running total and update that with the square of the most recent sample. To avoid your running total from growing and growing over time, add an exponential decay. In pseudocode:
decay_constant=0.999; // Some suitable value smaller than 1
total=0;
for t=1,...
// Exponential decay
total=total*decay_constant;
// Add in latest sample
total+=current_sample;
if total>threshold
// do something
end
end
Of course, you'll have to tune the decay constant and threshold to suit your application. If this isn't fast enough to run in real time, you have a seriously underpowered DSP...
You might try calculating two simple "statistics" - first would be spread (max-min). Silence will have very low spread. Second would be variety - divide the range of possible values into say 16 brackets (= value range) and as you go through the elements, determine in which bracket that element goes. Noise will have similar numbers for all brackets, whereas music or speech should prefer some of them while neglecting others.
This should be possible to do in just one pass through the array and you do not need complicated arithmetics, just some addition and comparison of values.
Also consider some approximation, for example take only each fourth value, thus reducing the number of checked elements to 80. For audio signal, this should be okay.
I did something like this a while back. After some experimentation I arrived at a solution that worked sufficiently well in my case.
I used the rate of change in the cube of the running average over about 120ms. When there is silence (only noise that is) the expression should be hovering around zero. As soon as the rate starts increasing over a couple of runs, you probably have some action going on.
rate = cur_avg^3 - prev_avg^3
I used a cube because the square just wasn't agressive enough. If the cube is to slow for you, try using the square and a bitshift instead. Hope this helps.
It is clearly that the complexity should be at least O(n). Probably some simple algorithms that calculate some value range are good for the moment but I would look for Voice Activity Detection on web and for related code samples.
I want to make a program that would record audio data using PortAudio (I have this part done) and then display the frequency information of that recorded audio (for now, I'd like to display the average frequency of each of the group of samples as they come in).
From some research I've done, I know that I need to do an FFT. So I googled for a library to do that, in C, and found FFTW.
However, now I am a little lost. What exactly am I supposed to do with the samples I recorded to extract some frequency information from them? What kind of FFT should I use (I assume I'd need a real data 1D?)?
And once I'd do the FFT, how do I get the frequency information from the data it gives me?
EDIT : I now found also the autocorrelation algorithm. Is it better? Simpler?
Thanks a lot in advance, and sorry, I have absolutely no experience if this. I hope it makes at least a little sense.
To convert your audio samples to a power spectrum:
if your audio data is integer data then convert it to floating point
pick an FFT size (e.g. N=1024)
apply a window function to N samples of your data (e.g. Hanning)
use a real-to-complex FFT of size N to generate frequency domain data
calculate the magnitude of your complex frequency domain data (magnitude = sqrt(re^2 + im^2))
optionally convert magnitude to a log scale (dB) (magnitude_dB = 20*log10(magnitude))