I want to create some simple heart rate monitor in LabVIEW.
I have sensor which gives me heart workflow (upper graph): Waveform
On second graph (lower graph) is amount of hills (0 - valley, 1 - hill) and that hills are heart beats (that is voltage waveform). From this I want to get amount of those hills, then multiply this number by 6 and I'll get heart rate per minute.
Measuring card I use: NI USB-6009.
Any idea how to do that?
I can sent a VI file if anyone will be able to help me.
You could use Threshold Peak Detector VI
This VI does not identify the locations or the amplitudes of peaks
with great accuracy, but the VI does give an idea of where and how
often a signal crosses above a certain threshold value.
You could also use Waveform Peak Detection VI
The Waveform Peak Detection VI operates like the array-based Peak
Detector VI. The difference is that this VI's input is a waveform data
type, and the VI has error cluster input and output terminals.
Locations displays the output array of the peaks or valleys, which is
still in terms of the indices of the input waveform. For example, if
one element of Locations is 100, that means that there is a peak or
valley located at index 100 in the data array of the input waveform.
Figure 6 shows you a method for determining the times at which peaks
or valleys occur.
NI have a great tutorial that should answer all your questions, it can be found here:
I had some fun recreating some of your exercise here. I simulated a squarewave. In my sample of the square wave, I know how many samples I have and the sampling frequency. As a result, I calculate how much time my data sample represents. I then count the number of positive edges in the sample. I do some division to calculate beats/second and multiplication for beats/minute. The sampling frequency, Fs, and number of samples, N or #s are required to calculate your beats per minute metric. Their uses are shown below.
The contrived VI
Does that lead you to a solution for your application?
Related
At first blush this presumably means -
(1) looking only at lower IR frequencies,
(2) select a IR frequency cut-off for low frequency buckets of the u/v FFT grid
(3) Once we have that, derive the power distribution - squares of amplitudes - for that IR range of frequency buckets the camera supports.
(4) Fit that distribution against the Rayleigh-Jones classical Black Box radiation formula:
(https://en.wikipedia.org/wiki/Rayleigh%E2%80%93Jeans_law#Other_forms_of_Rayleigh%E2%80%93Jeans_law)
(5) Assign a Temperature of 'best fit'.
The units for B(ν,T) are Power per unit frequency per unit surface area at equilibrium Temperature
Of course, this leaves many details out, such as (6) cancelling background, etc, but one could perhaps use the opposite facing camera to assist in that. Where buckets do not straddle the temperature of interest, (7) use a one-sided distribution to derive an inferred Gaussian curve to fit the Rayleigh-Jeans curve at that derived central frequency ν, for measured temperature T.
Finally (8) check if this procedure can consistently detect a high vs low surface temperature (9) check if it can consistently identify a 'fever' temperature (say, 101 Fahrenheit / 38 Celcius) pointing at a forehead.
If all that can be done, (10) Voila! a body fever detector
So those who are capable can fill us in on whether this is possible to do so for eventual posting at an app store as a free Covid19 safe body temperature app? I have a strong sense there's quite a few out there who can verify this in a week or two!
It appears that the analog signal assumed in (1) and (2) are not available in the Android digital Camera2 interface.
Android RAW image stream, that is uncompressed YUV, is already encoded Y green monochrome, and U,V are blue and red shifts from zero for converting green monochrome to color.
The original analog frequency / energy signal is not immediately accessible. So adaptation is not possible (yet).
To my understanding, the noise floor for each USRP may be different. I want to know how I can calculate the noise floor without physically going into the fft and spotting it out manually. I want to know if there is a block in GNU Radio that will calculate this, or if there is a stream of blocks I can use to calculate it. Please provide a block diagram in your answer ( block 1 ---> block 2 ---> ...etc.).
For my application, let's say I have a QT GUI frequency sink that is showing all noise at the moment. I want to calculate the noise floor so that I have a value that represents "no signal present" ie. noise. Once I have this value, I plan to set a threshold 5dB higher than the noise floor to indicate that a signal has been detected. I've been able to kind of eye ball the average noise value from the QT GUI Frequency Sink but that's not good enough for me. I want to be able to calculate it and not have to look into the plot every time to update the noise value every time I change USRPs.
For instance:
You can see the average noise value for this is around -55dB. I want to calculate this without having to eye ball it. This way, when a signal gets transmitted at (in this example) 0Hz, then the power of the signal will increase and I can see if a signal was detected.
I need to calculate the FFT of audiodata in an Android Project and I use jTransforms to achieve this.
The samples of the audiodata are a few seconds long and are recorded with a samplerate of 11025 Hertz.
I am not sure how to set the length of the FFT in jTransforms.
I do not really need high frequency resolution, so a size of 1024 would be enough.
But from what I have understood learning about the FFT, if I decrease the FFT size F and use a sample with a lenght of N > F, only the first F values of the original sample are transformed.
Is that true or did I understand something wrong?
If it is true, is there an efficient way to tranform the whole signal and decreasing the FFT-Size afterwards?
I need this to classify different signals using Support Vector Machines, and FFT-Sizes > 1024 would give me too much features as output, so I would have to reduce the result of the FFT to a more compact vector.
If you only want the FFT magnitude results, then use the FFT repeatedly on successive 1024 chunk lengths of data, and vector sum all the successive magnitude results to get an estimate for the entire much longer signal.
See Welch's Method on estimating spectral density for an explanation of why this might be a useful technique.
Im not familiar with the jTransform library, but do you really set the size of the transform before calculating it? Amplitude values of the time-domain signal and the sampling frequency (11.025 kHz) is enough to calculate the FFT (note that the FFT assumes constant sampling rate)
The resolution in frequency domain will be determined by Nyquist's theorem; the maximum resolvable frequency in your signal will be equal to half your sampling rate. In other words, sampling your signal with 11.025 kHz, you can expect your frequency graph to contain frequency values (and corresponding amplitudes) between 0 Hz - 5.5125 kHz.
UPDATE:
The resolution of the FFT (the narrowness of the frequency bins) will increase/improve if your input signal is longer, thus 1024 samples might not be a long sequence enough if you need to distinguish between very small changes in frequency. If thats not a problem for you application, and the nature of your data is not variying quickly, and you have the processing time, then taking an average of 3-4 FFT estimates will greatly reduce noise and improve estimates.
Have done fft (see earlier posting if you are interested!) and got a result, which helps me. Would like to analyse the noisiness / spikiness of an array (actually a vb.nre collection of single). Um, how to explain ...
When signal is good, fft power results is 512 data points (frequency buckets) with low values in all but maybe 2 or 3 array entries, and a decent range (i.e. the peak is high, relative to the noise value in the nearly empty buckets. So when graphed, we have a nice big spike in the values in those few buckets.
When signal is poor/noisy, data values spread (max to min) is low, and there's proportionally higher noise in many more buckets.
What's a good, computationally non-intensive was of analysing the noisiness of this data set? Would some kind of statistical method, standard deviations or something help ?
The key is defining what is noise and what is signal, for which modelling assumptions must be made. Often an assumption is made of white noise (constant power per frequency band) or noise of some other power spectrum, and that model is fitted to the data. The signal to noise ratio can then be used to measure the amount of noise.
Fitting a noise model depends on the nature of your data: if you know that the real signal will have no power in the high frequency components, you can look there for an indication of the noise level, and use the model to predict what the noise will be at the lower frequency components where there is both signal and noise. Alternatively, if your signal is constant in time, taking multiple FFTs at different points in time and comparing them to get a standard deviation for each frequency band can give the level of noise present.
I hope I'm not patronising you to mention the issues inherent with windowing functions when performing FFTs: these can have the effect of introducing spurious "noise" into the frequency spectrum which is in fact an artifact of the periodic nature of the FFT. There's a tradeoff between getting sharp peaks and 'sideband' noise - more here www.ee.iitm.ac.in/~nitin/_media/ee462/fftwindows.pdf
Calculate a standard deviation and then you decide the threshold that will indicate noise. In practice this is usually easy and allows you to easily tweak the "noise level" as needed.
There is a nice single pass stddev algorithm in Knuth. Here is link that describes an implementation.
Standard Deviation
calculate the signal to noise ratio
http://en.wikipedia.org/wiki/Signal-to-noise_ratio
you could also check the stdev for each point and if it's under some level you choose then the signal is good else it's not.
wouldn't the spike be
treated as a noise glitch in SNR, an
outlier to be discarded, as it were?
If it's clear from the time-domain data that there are such spikes, then they will certainly create a lot of noise in the frequency spectrum. Chosing to ignore them is a good idea, but unfortunately the FFT can't accept data with 'holes' in it where the spikes have been removed. There are two techniques to get around this. The 'dirty trick' method is to set the outlier sample to be the average of the two samples on either site, and compute the FFT with a full set of data.
The harder but more-correct method is to use a Lomb Normalised Periodogram (see the book 'Numerical Recipes' by W.H.Press et al.), which does a similar job to the FFT but can cope with missing data properly.
Trying to understand an fft (Fast Fourier Transform) routine I'm using (stealing)(recycling)
Input is an array of 512 data points which are a sample waveform.
Test data is generated into this array. fft transforms this array into frequency domain.
Trying to understand relationship between freq, period, sample rate and position in fft array. I'll illustrate with examples:
========================================
Sample rate is 1000 samples/s.
Generate a set of samples at 10Hz.
Input array has peak values at arr(28), arr(128), arr(228) ...
period = 100 sample points
peak value in fft array is at index 6 (excluding a huge value at 0)
========================================
Sample rate is 8000 samples/s
Generate set of samples at 440Hz
Input array peak values include arr(7), arr(25), arr(43), arr(61) ...
period = 18 sample points
peak value in fft array is at index 29 (excluding a huge value at 0)
========================================
How do I relate the index of the peak in the fft array to frequency ?
If you ignore the imaginary part, the frequency distribution is linear across bins:
Frequency#i = (Sampling rate/2)*(i/Nbins).
So for your first example, assumming you had 256 bins, the largest bin corresponds to a frequency of 1000/2 * 6/256 = 11.7 Hz.
Since your input was 10Hz, I'd guess that bin 5 (9.7Hz) also had a big component.
To get better accuracy, you need to take more samples, to get smaller bins.
Your second example gives 8000/2*29/256 = 453Hz. Again, close, but you need more bins.
Your resolution here is only 4000/256 = 15.6Hz.
It would be helpful if you were to provide your sample dataset.
My guess would be that you have what are called sampling artifacts. The strong signal at DC ( frequency 0 ) suggests that this is the case.
You should always ensure that the average value in your input data is zero - find the average and subtract it from each sample point before invoking the fft is good practice.
Along the same lines, you have to be careful about the sampling window artifact. It is important that the first and last data point are close to zero because otherwise the "step" from outside to inside the sampling window has the effect of injecting a whole lot of energy at different frequencies.
The bottom line is that doing an fft analysis requires more care than simply recycling a fft routine found somewhere.
Here are the first 100 sample points of a 10Hz signal as described in the question, massaged to avoid sampling artifacts
> sinx[1:100]
[1] 0.000000e+00 6.279052e-02 1.253332e-01 1.873813e-01 2.486899e-01 3.090170e-01 3.681246e-01 4.257793e-01 4.817537e-01 5.358268e-01
[11] 5.877853e-01 6.374240e-01 6.845471e-01 7.289686e-01 7.705132e-01 8.090170e-01 8.443279e-01 8.763067e-01 9.048271e-01 9.297765e-01
[21] 9.510565e-01 9.685832e-01 9.822873e-01 9.921147e-01 9.980267e-01 1.000000e+00 9.980267e-01 9.921147e-01 9.822873e-01 9.685832e-01
[31] 9.510565e-01 9.297765e-01 9.048271e-01 8.763067e-01 8.443279e-01 8.090170e-01 7.705132e-01 7.289686e-01 6.845471e-01 6.374240e-01
[41] 5.877853e-01 5.358268e-01 4.817537e-01 4.257793e-01 3.681246e-01 3.090170e-01 2.486899e-01 1.873813e-01 1.253332e-01 6.279052e-02
[51] -2.542075e-15 -6.279052e-02 -1.253332e-01 -1.873813e-01 -2.486899e-01 -3.090170e-01 -3.681246e-01 -4.257793e-01 -4.817537e-01 -5.358268e-01
[61] -5.877853e-01 -6.374240e-01 -6.845471e-01 -7.289686e-01 -7.705132e-01 -8.090170e-01 -8.443279e-01 -8.763067e-01 -9.048271e-01 -9.297765e-01
[71] -9.510565e-01 -9.685832e-01 -9.822873e-01 -9.921147e-01 -9.980267e-01 -1.000000e+00 -9.980267e-01 -9.921147e-01 -9.822873e-01 -9.685832e-01
[81] -9.510565e-01 -9.297765e-01 -9.048271e-01 -8.763067e-01 -8.443279e-01 -8.090170e-01 -7.705132e-01 -7.289686e-01 -6.845471e-01 -6.374240e-01
[91] -5.877853e-01 -5.358268e-01 -4.817537e-01 -4.257793e-01 -3.681246e-01 -3.090170e-01 -2.486899e-01 -1.873813e-01 -1.253332e-01 -6.279052e-02
And here is the resulting absolute values of the fft frequency domain
[1] 7.160038e-13 1.008741e-01 2.080408e-01 3.291725e-01 4.753899e-01 6.653660e-01 9.352601e-01 1.368212e+00 2.211653e+00 4.691243e+00 5.001674e+02
[12] 5.293086e+00 2.742218e+00 1.891330e+00 1.462830e+00 1.203175e+00 1.028079e+00 9.014559e-01 8.052577e-01 7.294489e-01
I'm a little rusty too on math and signal processing but with the additional info I can give it a shot.
If you want to know the signal energy per bin you need the magnitude of the complex output. So just looking at the real output is not enough. Even when the input is only real numbers. For every bin the magnitude of the output is sqrt(real^2 + imag^2), just like pythagoras :-)
bins 0 to 449 are positive frequencies from 0 Hz to 500 Hz. bins 500 to 1000 are negative frequencies and should be the same as the positive for a real signal. If you process one buffer every second frequencies and array indices line up nicely. So the peak at index 6 corresponds with 6Hz so that's a bit strange. This might be because you're only looking at the real output data and the real and imaginary data combine to give an expected peak at index 10. The frequencies should map linearly to the bins.
The peaks at 0 indicates a DC offset.
It's been some time since I've done FFT's but here's what I remember
FFT usually takes complex numbers as input and output. So I'm not really sure how the real and imaginary part of the input and output map to the arrays.
I don't really understand what you're doing. In the first example you say you process sample buffers at 10Hz for a sample rate of 1000 Hz? So you should have 10 buffers per second with 100 samples each. I don't get how your input array can be at least 228 samples long.
Usually the first half of the output buffer are frequency bins from 0 frequency (=dc offset) to 1/2 sample rate. and the 2nd half are negative frequencies. if your input is only real data with 0 for the imaginary signal positive and negative frequencies are the same. The relationship of real/imaginary signal on the output contains phase information from your input signal.
The frequency for bin i is i * (samplerate / n), where n is the number of samples in the FFT's input window.
If you're handling audio, since pitch is proportional to log of frequency, the pitch resolution of the bins increases as the frequency does -- it's hard to resolve low frequency signals accurately. To do so you need to use larger FFT windows, which reduces time resolution. There is a tradeoff of frequency against time resolution for a given sample rate.
You mention a bin with a large value at 0 -- this is the bin with frequency 0, i.e. the DC component. If this is large, then presumably your values are generally positive. Bin n/2 (in your case 256) is the Nyquist frequency, half the sample rate, which is the highest frequency that can be resolved in the sampled signal at this rate.
If the signal is real, then bins n/2+1 to n-1 will contain the complex conjugates of bins n/2-1 to 1, respectively. The DC value only appears once.
The samples are, as others have said, equally spaced in the frequency domain (not logarithmic).
For example 1, you should get this:
alt text http://home.comcast.net/~kootsoop/images/SINE1.jpg
For the other example you should get
alt text http://home.comcast.net/~kootsoop/images/SINE2.jpg
So your answers both appear to be correct regarding the peak location.
What I'm not getting is the large DC component. Are you sure you are generating a sine wave as the input? Does the input go negative? For a sinewave, the DC should be close to zero provided you get enough cycles.
Another avenue is to craft a Goertzel's Algorithm of each note center frequency you are looking for.
Once you get one implementation of the algorithm working you can make it such that it takes parameters to set it's center frequency. With that you could easily run 88 of them or what ever you need in a collection and scan for the peak value.
The Goertzel Algorithm is basically a single bin FFT. Using this method you can place your bins logarithmically as musical notes naturally go.
Some pseudo code from Wikipedia:
s_prev = 0
s_prev2 = 0
coeff = 2*cos(2*PI*normalized_frequency);
for each sample, x[n],
s = x[n] + coeff*s_prev - s_prev2;
s_prev2 = s_prev;
s_prev = s;
end
power = s_prev2*s_prev2 + s_prev*s_prev - coeff*s_prev2*s_prev;
The two variables representing the previous two samples are maintained for the next iteration. This can be then used in a streaming application. I thinks perhaps the power calculation should be inside the loop as well. (However it is not depicted as such in the Wiki article.)
In the tone detection case there would be 88 different coeficients, 88 pairs of previous samples and would result in 88 power output samples indicating the relative level in that frequency bin.
WaveyDavey says that he's capturing sound from a mic, thru the audio hardware of his computer, BUT that his results are not zero-centered. This sounds like a problem with the hardware. It SHOULD BE zero-centered.
When the room is quiet, the stream of values coming from the sound API should be very close to 0 amplitude, with slight +- variations for ambient noise. If a vibratory sound is present in the room (e.g. a piano, a flute, a voice) the data stream should show a fundamentally sinusoidal-based wave that goes both positive and negative, and averages near zero. If this is not the case, the system has some funk going on!
-Rick