Robert Wolfe (@Robert Wolfe)
I've done 2) above, but it was for a DSP processor. Found an online C version of the filter design, and just ran it each time a new set of filter configuration...
this recent thread discusses some of the potential perils of assuming that a FFT modification, will inverse back to the time domain as desired/assumed:filtering...
Confused a bit by this. From all information read, and replies here, zero'ing the FFT bin appears to only be a brick wall for the signal component associated with...
The original question was about the impact on filtering. I'd think the criteria would be accuracy, relative to time domain convolution, or it's equivalent multiplication...
The thread here appears to speak to some of the shortcomings of this approach:matlab - Why is it a bad idea to filter by zeroing out FFT bins? - Signal Processing...
Seems application dependent. But if you're going to do the 16384 FFT, wouldn't it be best with actual data (collect enough packets). Else you're processing the...
Those are ancient. Did quite a bit of C32 myself, many moons ago. Maybe time to update the platform :) I'd guess the White Mountain parts are not compatible...
If the Python filter looked great, I would continue to refine it to more closely match your real-time implementation, until you see where the issues are encountered. ...
Did you intend to include expressions?, that you're asking about. I don't see them in your post (but could just be me).Regards,Robert
Think of the biquad definition:( a0 * y[n] ) + ( a1 * y[ n - 1 ] ) + ( a2 * y[ n - 2 ] ) = ( b0 * x[n] ) + ( b1 * x[ n - 1 ] ) + ( b2 * x[ n - 2 ] )So, with a0...
Welcome. In this case, I did (process all biquads first). But in other cases, I did the check for saturation/overflow after each of the biquad MAC operation...
Hi,Here's an example from a filter simulation, with 16 bit coefficients, where the round and saturation check is done after each biquad.Robertiir_16.c
Hello,Sorry, I don't remember an app note (but very well could have existed). But I had done exactly what you mentioned, with C55 processors - bit accurate C simulations...
You want to filter to at least ( 1 MHz / 128 ), which is the Nyquist of your new sample rate, to avoid aliasing. Yes, it could be a long FIR filter, but you only...
I guess with z = sqrt( ( I * I ) + ( Q * Q ) ), as Q becomes <<< I, the contribution from Q becomes less and less, and the result approaches sqrt( I *...
Hello,I believe the pspectrum (power spectrum) is probably doing the following:pspectrum[ n ] = ( 2 / # of FFT points ) * sqrt( ( fft.real[ n ] * fft_real[ n ]...
If all else fails, my tried and true method of implementing (and debugging) filters on embedded devices, is:1) first, code the filter up in "offline C" on a PC (GNU...
Another idea is to examine all configuration items for the signal generation, acquisition, algorithm and overall processing, with a fine tooth comb. Review, and...
Hello,I don't know enough about SDR to help specifically with algorithm issues. And it seems like there are a lot of processing steps, that could introduce/contribute...
Hello,Doesn't a 65 MHz signal require a sample rate of at least 130 MHz, to avoid aliasing?, per Nyquist point.Regards,Robert
Hello,If you plan to sit behind a computer all day, and write MATLAB/Simulink algorithms, probably not as much a need for computer architecture. But if you ever...
You're always looking at the ratio of the largest coefficient to the smallest. Too big, and you start to get into problems, because after dividing down to get...
M is the number of input samples used, and N is the number of output frequency bins calculated. For a general FFT, M and N are the same, i.e. N values in time...
Let's say you want to generate 512 DFT outputs for every 4 samples input.M = 4N = 512log2( N ) = 94 < 9so sliding DFT can be computationally superior to traditional...
Hello,You've been given some good references, to which I'll add one more. It underscores the need for noise addition, in addition to oversampling (4X for each...
Bravo, good job. You have your problem resolved, and we have all learned something interesting in the process ;)Best,Robert
Hello,I'm not an expert on this. But just a brief google search came up with this post, with the most relevant part in the answer at the end, in my view. Seems...
Hello,If your only goal is to filter, unless it's at something like a 500 K sample rate, most anything will work. I personally would go arm core with FPU and...
It's hard to debug code, without digging in for a while. What I would recommend is creating a circular buffer from some memory block (largest you have free). ...
Hello,You will need to convert the .out file to binary format, and flash it to your board. There are tools, and instructions for that. Start here:http://software-dl.ti.com/trainingTTO/trainingTTO_...You...
Probably replicating all the fine answers already, but when considering these types of situations, I always remember my golden rule of spectrum analysis: bin...
I'm not necessarily an expert in these matters, but you'll need to design your bandpass to give sufficient attenuation at +/- 30 Hz, to take out the 70 Hz. Typically...
Depends on the filter cutoff. Let's call Fa the square wave fundamental frequency. If the filter cutoff is between Fa and 2xFa, you'll get a sinusoid back with...
By examination, you can see that the two components making up v(n), i.e. arrows into the summing point, are the input x(n) with the sum of:1) product of -a1 and...
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