Building a Digital Filter: How it works + Simulation + Example

Everything important you need to know to start designing your own digital filters. Thank you very much C. Daniel Boschen.
Links:
- Daniel's Linked In: / danboschen
- FEDEVEL courses: fedevel.com/courses
Links to Dan Boschen's Courses:
ieeeboston.org/courses/
www.dsprelated.com/courses
Other relevant links:
Microchip MPLAB: www.microchip.com/en-us/tools...
Anaconda Distribution (includes Python tools and Jupyter Notebook): www.anaconda.com/download
SAM E51 Curiosity Nano Evaluation Kit: www.microchip.com/en-us/devel...
PMOD I2S2 ADC/DAC: digilent.com/reference/pmod/p...
Music used in the demo:
ISRC: USUAN1900041
Uploaded: 2019-12-18
credits:
"Adding the Sun" Kevin MacLeod (incompetech.com)
Licensed under Creative Commons: By Attribution 4.0 License
creativecommons.org/licenses/b...
Chapters:
00:00 What is this video about
00:40 Our example explained
04:19 How filters work
07:48 What is a digital filter
10:46 Basic elements of digital filters
13:08 How filters manipulate with signal
15:00 Feed-forward (FIR) filter
19:39 Feed-back (IIR) filter
23:36 Biquad - Second Order Section (SOS)
28:49 The hardware we used
33:09 DMA explained
42:30 Audio file before filtering
44:16 Calculating filter coefficients
47:03 Our filter frequency response
48:59 Audio file after filtering
49:59 Explained how our filter works
54:52 C code of our filter
57:49 Daniel's courses
1:00:35 Setup for simulation - Jupyter and anaconda
1:05:14 Simulation - Initialization
1:10:38 Simulation - Showing the spectrum
1:13:04 Simulation - The filter
1:22:35 Simulation - Filtering audio file
1:25:47 Simulation - C code of filter
1:28:06 Explaining the code in microcontroller
1:34:33 The filter in C code of microcontroller
1:37:50 How to do this by yourself
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Would you like to support me? It's simple:
- Sign up for online courses hosted on our platform: fedevel.com/
- You can also support me through Patreon: / robertferanec
- FEDEVEL Discord: / discord
It is much appreciated. Thank you,
- Robert

[@iskogproductions6728]

I had tens of hours of digital signal processing and signal filtering lectures, but this is the most intuitive course I've come across. Thank you for the clear explanations.

[@MrLRankin2]

Finally! Someone that can actually explain digital filters so that we mere humans can actually understand.

[@robegatt]

And enthusiastically does so !

[@geofflethbridge8732]

I think that was a great presentation of DSP, and it was great to see your enthusiasm for the subject and teaching. :)

[@thomhpl]

Nice demo! I did something very similar a few days ago to add some low pass filtering for a speed sensor in a motor control loop firmware, using a jupyter notebook as well to design a simple FIR filter. I actually then used python to directly generate C code containing the array with the filter coefficients.

[@videogenie1236]

Just watched the presentation of the two PCB's. I like the dev kit pcb"s and ordered them for delivery tomorrow from RS Components in the UK. 🇬🇧 Thank-you both

[@rosskappa5410]

God bless people like you.

[@muhdiversity7409]

Excellent content, Robert. Thank you so much for sharing.

[@em2129]

Thank you! This is the best video I have seen on this topic so far!

[@ketankothari247]

Beautifully explained sir 👏 👌 ❤️

[@benish0r]

Very cool video and Dan is an absolutely amazing teacher. I should know, as I am one of his students :-)

[@BigA1]

The subject material is very well presented - thank you.

[@hmpeter]

Excellent video, thank you guys! 😀

[@lacyrjoao2553]

Your blog are great and you're great, great, great.

[@MusicBent]

Digital signal processing was one of my favorite college courses. The math was tough, but learning how to make practical IIR and FIR filters was a great payoff. Audio is a great way to experiment because you can experience it. Most of my work nowadays is in RF / software defined radio processing which I also find fascinating

[@user-uk5ep9hm5k]

It was very informative. Thanks a lot.

[@Sanjeev...]

Good dsp presentation thanks

[@AnalogDude_]

Cool video, the low and hi pass filter? Cool that digilent PCB, Mouser has it.

[@ziegelhof]

Dakujem za velmi uzitocne video

[@jeanclaudejosephbadji5418]

Just amazing!

[@jeanclaudejosephbadji5418]

Thank you very much

[@youtubeaccount931]

Fantastic

[@landspide]

Probably get away with software fpu in this case, not sure how fast the processor but would be surprised if it couldn't keep up. Very good presentation for beginners though, impressive!!!

[@robegatt]

120 Mhz means at that sample frequency it is little more than 1000 machine code instructions per channel to do the filtering... not so sure it would fit a software fp.

[@landspide]

@@robegatt yeah good point, one would have to try to be sure..

[@expertis307]

Excelente !!!!

[@kabandajamir9844]

So nice thanks sir

[@rabie-et8zg]

the problem of delay you can remove it by using a zero phase shift filters , great video Robert

[@danboschen9613]

This is a real time filter and zero-phase filters are non-causal (so will only work with post processing and can’t be applied here).

[@robertpeters9438]

It would be great to have a video delay to match the audio filter delay.

[@ebubekiralparslancaliskan3985]

Thank you very much for the video, but I want you to add Turkish subtitles, I am looking forward to your feedback

[@xxportalxx.]

One thing I wish was brought up: layering stages. For instance in the 1khz tone example he showed a tighter filter takes longer to build, would there be an advantage to layering a looser filter with a tight filter to decrease that initial blip while still getting that tight rejection? Or would the loose filter make the tight filter redudndant? If you wanted to reject multiple bands do the two stages add delay independently, or is the delay the same as if they were applied individually? Do you add these stages to do so, or is there a means to merge them into a smaller structure?

[@danboschen9613]

Great questions and thoughts @xxportalxx !! Cascading multiple second order sections (one after the other if that is what you mean by layering) is the common approach to implement higher order filters. The big advantage in doing this is isolating the filter poles from interacting with each other leading to higher stability implementations, especially when the poles are close to the frequency axis ("higher Q" resonators). That's the primary advantage, and the delay and performance is predicted by the prototype higher order filter. If we used that approach to isolate two different frequencies as the notch filter demonstrated here, the delay (which is only in vicinity of the notch and creates the start-up transient) is as it would be for each filter individually, as long as they are separated sufficiently to not interact with each other. The deeper details are beyond the scope of this short video or what I could fairly answer in a comment, but it is something I go into as part of the "DSP for Wireless Communications" course referenced in the link (where we have much more time and the right context to explain). That course is coming up next month!

[@xxportalxx.]

@danboschen9613 Amazing presentation, and thank you for your response. DSP is generally presented as a black box above us mere mortals, very eye opening demonstration that it doesn't have to be out of reach.

[@biswajit681]

Magnetics design for SMPS please

[@fas9600]

I understand there are specialized courses, but it would be great if you could share the demo jupyter notebook. thanks.

[@biiber]

some MCU vendors (like ST) have dedicated hardware (FMAC) to do these calculations in fixed point numbers. so the DMA could stream directly to and from the FMAC and the CPU can do other things or sleep. just want to point out how solid this concept is that it is implemented in hardware.

[@25.muhammadhilmihaidary60]

wow

[@chinametal]

The algorithm presented in 54:52 will not work.....the variables must retain the previous information, so they cannot be reset every time they enter in the function. They have to be outside, that is, be global and not local. I think that's it..... the programmers can confirm it or not!

[@danboschen9613]

@chinametal they are defined as “static variables” which means they retain their state from call to call.

[@robegatt]

Optimized version would not pass coefficients everytime.

[@Jindraxx20]

3rd to view 1st to like the video (>.

[@RobertFeranec]

The first 2 views were mine (before publishing) so you are the first one to see it after publishing :)

[@ulbed]

You can hear the 1 khz tone in the background all the time. Very annoying. Now I need to build a filter before watching this video.

[@danboschen9613]

Ha ha! I am 1 KHz tone deaf from all the testing I did

[@outgoingbot]

Professor was on some BS when asked where his notch filter coefficients came from.

[@lakwejh]

Wish I had my teachers explained this clearly......on the other hand may be I should have paid more attention