Robert Wolfe (@Robert Wolfe)

20+ years DSP, embedded systems and firmware. Music nut.

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 *...

Re: Dr. Mark Wickert's scipy-dsp-comm Work

Reply posted 2 months ago (10/20/2022)
Thanks.  Will add this to the list of excellent instructional information out there.Robert

Re: Difference between fft and pspectrum in Matlab

Reply posted 3 months ago (09/19/2022)
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 ]...

Re: Having problem in debugging an issue with IIR filter

Reply posted 5 months ago (07/14/2022)
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...

Re: Weird Signals when acquiring with SDR boards

Reply posted 5 months ago (07/06/2022)
Another idea is to examine all configuration items for the signal generation, acquisition, algorithm and overall processing, with a fine tooth comb.  Review, and...

Re: Weird Signals when acquiring with SDR boards

Reply posted 5 months ago (07/06/2022)
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

Re: How much CompE for DSP?

Reply posted 6 months ago (05/25/2022)
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...

Re: doubt regarding the sentence in sliding dft

Reply posted 11 months ago (01/05/2022)
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...

Re: doubt regarding the sentence in sliding dft

Reply posted 11 months ago (12/29/2021)
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...

Re: Free book: Software-Defined Radio for Engineers

Reply posted 12 months ago (12/08/2021)
Got it, thanks. Robert

Re: Using noise to increase resolution of ADC

Reply posted 1 year ago (11/22/2021)
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...

Re: TAS3204 DSP Audio Processing

Reply posted 1 year ago (11/19/2021)
You can start at the TI E2E forums:
Bravo, good job.  You have your problem resolved, and we have all learned something interesting in the process ;)Best,Robert
Hao,Sounds good, please do share if you are able to generate more rigorous proof.Regards,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...

Re: Audio Signal Filter

Reply posted 1 year ago (08/15/2021)
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...

Re: ARM IIR Filter - Why No DAC Output?

Reply posted 1 year ago (07/16/2021)
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). ...

Re: Fundamental CCS Ti C67x questions

Reply posted 1 year ago (07/09/2021)
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:

Re: FFT spectrum shift after time domain decimation

Reply posted 1 year ago (06/15/2021)
Probably replicating all the fine answers already, but when considering these types of situations, I always remember my golden rule of spectrum analysis:    bin...

Re: Overdriven Sine Wave through DSP Filter

Reply posted 2 years ago (05/28/2021)
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...

Re: Overdriven Sine Wave through DSP Filter

Reply posted 2 years ago (05/28/2021)
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...

Re: Direct Form II

Reply posted 2 years ago (05/27/2021)
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...

Re: How to test my FFT implementation?

Reply posted 2 years ago (03/07/2021)
Can use sine data (as drmike suggested).  Here's an implementation for filling input data to an FFT, to place a single peak in the bin and amplitude of your choosing: ...

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