I'm a long time user of DSPs and DSP algorithms in active noise and vibration control. More recently, I have been using faster microcontrollers to do what I need.

Re: FFT spectrum shift after time domain decimation

Reply posted 6 months ago (06/15/2021)
I'm just curious how someone ends up with a 44800 Hz sample rate.  :)

Re: Moving Sound Source Database for SSL

Reply posted 6 months ago (06/13/2021)
It's not my area and I wasn't/aren't even sure if you were searching for binaural recordings or data recoded using a mic array. If binaural, I was originally...

Re: Overdriven Sine Wave through DSP Filter

Reply posted 6 months ago (05/30/2021)
Got it.  My thoughts immediately go to using a lock-in amplifier.  ...but noise interference is always better avoided than grappled with. fred is explaining it...

Re: Filter approach for helicopter application

Reply posted 6 months ago (05/29/2021)
Got it to fs = 100 Hz.  In addition to the other comments, I want to offer this for thought...If your ADC is sampling at 100 Hz, it likely has an antialiasing...

Re: Filter approach for helicopter application

Reply posted 6 months ago (05/28/2021)
What is your sample rate?  I couldn't ascertain it from the information above.

Re: Overdriven Sine Wave through DSP Filter

Reply posted 6 months ago (05/28/2021)
So, are you only interested in a 100Hz tone?  ...or a band around 100Hz?  ...and, if you are only interested in the 100Hz tone, what about that tone is of interest...

Re: Direct Form II

Reply posted 6 months ago (05/27/2021)
To add to the good replies... It also might be helpful intuitively to look at Direct Form I and see how it translates to Direct Form II.Given a 2nd order filter...Here...
There is good advice above.  I will add that I had the same question as you when first implementing adaptive active noise and vibration systems.  If the I/O was...

Re: Matlab, properly using IFFT, FIR Filter Desing

Reply posted 2 years ago (05/21/2020)
Hi Jim,You didn't miss anything.  That is a typo from the original MATLAB code posted by the OP.  All of the plot x-axes labels are set the same, inadvertently. ...

Re: PhD in E.E. is it really worth it

Reply posted 2 years ago (05/08/2020)
That's not a silly question, Marcin.First, a joke: When you think that you know everything, they give you a bachelors degree.  When you realize that you know nothing,...

Re: Computing the delay from a room response FIR filter

Reply posted 2 years ago (05/08/2020)
Late to the party...I prefer multisines.  A good classic paper on excitation sequences is this:J. Schoukens, R. Pintelon, E. van der Ouderna, J. Renneboog, “Survey...

Re: Undersampling FM radio frequencies

Reply posted 2 years ago (05/01/2020)
I didn't intend to suggest that the demodulation was to be skipped.  Only that I would not be trying to work on the entire band as a first go.

Re: Undersampling FM radio frequencies

Reply posted 2 years ago (04/30/2020)
The spacing is 200Khz and everything is supposed to exist within 75KHz of the band center frequency by declaration.  Really... nothing of interest should be outside...

Re: Matlab, properly using IFFT, FIR Filter Desing

Reply posted 2 years ago (04/27/2020)
You are welcome.  I teach at Valparaiso University.  ...a small school southeast of Chicago.

Re: Matlab, properly using IFFT, FIR Filter Desing

Reply posted 2 years ago (04/27/2020)
Hello AL,Sorry for delay in getting back here.  The university semester is ending and I do not have much spare time with writing exams and all. The reason for...

Re: Matlab, properly using IFFT, FIR Filter Desing

Reply posted 2 years ago (04/24/2020)
I think that your problem is the mismatch between your step through h in the ifft() and your actual frequency response - which (I think) is kinda what you said at...
Hi max,I understand better your question.  I made reference to "fuzzifying" your difference.  In an essence, you are applying a hard threshold to your difference...
Hi max,I'm not sure what your question has to do with existing IIR or EWMA approaches.  As near as I can figure... you are quantizing a subcomponent of your filter...
Hi Deepak,Yes, sorry.  I had forgotten about that constraint.  ...and yes to the parameters of the spectrogram.  I had noticed the sensitivity before I got off...
Hi Deepak,The issue looks to be that you are normalizing your signal magnitude at the end of your creation of your final additive block.  Please keep in mind that...
Hi Deepak,You are welcome.  Glad to read of your good result.  I will take a look at the code today.Best regards,Dan
One other thought, also, for someone willing to throw hardware at this problem...  Connect a random noise generator to the input of a high-order switch-capacitor...
Hi Deepak,  The variable sample rate idea of the other poster - at least the way that I interpreted it - was to have random noise at various sample rates (and...
I like the variable sample rate idea from the other poster.  That said... if you can handle a piece-wise steps in BW, I recommend using a multi-sine whose frequency...
I did not see in your first post that you stated that you did not know the unfiltered signal.  ...and, upon rereading your first post, I still don't see where you...
Late to the party.If you knew the filter order, I would think that it would be easy enough.  You mention a 180deg out-of-phase at the center frequency.  That says...
I have some of the same questions.  Taking a stab at what you want, though...I would probably window the data to reduce spreading and take a 192000 point fft. ...

Re: Setting up codec on ADSP 21489

Reply posted 2 years ago (10/22/2019)
This codec and many others are time division multiplexed.  ...meaning that the configuration registers are sent every sample with the outgoing data.  Then, any...

Re: FIR Convolution Compile Error

Reply posted 2 years ago (10/17/2019)
...

Re: Pool Ball Pendulum Animation in MATLAB

Reply posted 2 years ago (06/17/2019)
No problem.  ...and very little of it isn't your code!  Thanks again!I really didn't want to repost all of yours, but I couldn't figure out a more elegant way...

Re: Pool Ball Pendulum Animation in MATLAB

Reply posted 2 years ago (06/17/2019)
Very cool, Rick.Below you will find Rick's original function with 5 commands added that generate an mp4 video of his animation.I will try to attach the generated...
The confusion might be the attempt to use the MATLAB filter command which assumes static coefficients across the entire dataset.  If you implement the filter yourself...

Re: Is this time-domain aliasing?

Reply posted 3 years ago (05/28/2019)
I'm guessing that - after the initial calculations and string clamping to length - that this setup was tuned more like a harp than anything else.  ...with much...

Re: Somewhat Off Topic: Manipulating a PDF File.

Reply posted 3 years ago (05/15/2019)
I use PDF Shuffler in Ubuntu.  If you are looking for a Windows application, google "PDF split and merge."  I recall using an application of that name that worked...

Re: What's your favorite modern DSP chip?

Reply posted 3 years ago (05/06/2019)
When using non-DSPs for DSP work, high speed, peripherals, and low cost can cure a lot of ills. But, as stated, their use comes with some baggage.I've been riding...

Re: half and quarter sample meaning

Reply posted 3 years ago (04/28/2019)
I believe that the primary purpose is for more realistic motion.

Re: half and quarter sample meaning

Reply posted 3 years ago (04/24/2019)
From the paper whose link I posted:Fig. 8 a) Half-pel pixel values - (b) Quater-pel pixel values.The integer-pels are your streamed luma samples.  Once the half-pel...

Re: half and quarter sample meaning

Reply posted 3 years ago (04/23/2019)
It is interpolating.  It is spacial, though, in case that is any source of confusion to the OP.I'm guessing - or rather Google is - that the snippet came from here:http://www.uta.edu/faculty/krrao/dip/Courses/EE535...This...
In addition to the above, rather than try to do simulations directly in DSP hardware, you would probably enjoy the process more if you were experimenting in a numeric...
Regarding your approach, if Vrms is going to truly be Vrms, you will want to add a calibration step.  That said... it appears that you want to calculate your effective...

Re: Phase in a filter

Reply posted 3 years ago (04/10/2019)
I'm on my phone, so I can't give so great of a response, but here it goes...  Filters are realized by storing and releasing energy in controlled ways.  That means...

Re: Phase in a filter

Reply posted 3 years ago (04/10/2019)
The gain responses in series filters are accumulated multiplicatively point-wise by frequency.  The phase responses in series filters are accumulated additively...

Re: Time delay

Reply posted 3 years ago (04/04/2019)
Good suggestions, all.To build on point #2... a unity-gain magnitude wideband 20ms delay is best implemented (digitally, of course) as a simple delay line buffer. ...

Re: DSP Dev Kits for learning

Reply posted 3 years ago (03/29/2019)
The options presented are all good in their own way.  I'm going to show my age here...The interesting thing with "learning DSP," though, is that it really is best...

Re: LMS data self correlation problem

Reply posted 3 years ago (03/02/2019)
Based on your description (even the original one), it seemed as though either adding leakage or a dither signal would help.  Leakage is better.  Also, you shouldn't...

Re: LMS data self correlation problem

Reply posted 3 years ago (03/02/2019)
In the absence of an explanation of the meaning of the term "bananas" in the context of adaptive filtering, I'd say add leakage.

Re: Plotting panning law graph

Reply posted 3 years ago (02/16/2019)
Does MATLAB allow us to send two different signals to our left and right speakers? Not sure if this was answered below.  The MATLAB sound() and soundsc() commands...

Re: 50Hz and harmonics filtering

Reply posted 3 years ago (02/08/2019)
This is a response to Donald's post above.  I edited it to add some more information and - inexplicably - the post is jumping around!  Please mentally insert...

Re: 50Hz and harmonics filtering

Reply posted 3 years ago (02/07/2019)
I apologize. The "shielded cable algorithm" was a (bad) joke about trying to keep the signal from being contaminated in the first place by hardware methods.  I...

Re: 50Hz and harmonics filtering

Reply posted 3 years ago (02/07/2019)
The best algorithm for achieving what you want (of which I am aware) is called the "shielded cable algorithm." :)But to your question...At first... I wasn't sure...
This is not my area of expertise.  Some questions catch my interest, though, and I use them to learn something new to which I wouldn't otherwise be exposed.After...

Re: Understanding the Comb Filter Frequency Response

Reply posted 3 years ago (01/20/2019)
You may want to start with this here on the site:https://www.dsprelated.com/freebooks/filters/Analy...

Re: Matlab-implementation of dynamic range compressor

Reply posted 3 years ago (01/16/2019)
Your question is: Can I compute the scaling factor just according to the block diagram, as I did in line 72 ?The answer is yes to line 72.  ...but your naming...

Re: Matlab-implementation of dynamic range compressor

Reply posted 3 years ago (01/14/2019)
For the second line of code that I called out, I was trying to make a two distinct points.  I'll state them again - both differently and in swapped order - in the...

Re: Matlab-implementation of dynamic range compressor

Reply posted 3 years ago (01/14/2019)
I'm just going to look at the diagram and describing everything - some obvious, some maybe not...The mapping between input level and output level is being done...

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