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

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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
- PCB Simulation vs. Real Board - DC Currents and Voltage drop • PCB Simulation vs. Rea...
- 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
------------------------------------------------------
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

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1 июн 2024

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Комментарии : 46

@iskogproductions6728 11 дней назад

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 17 дней назад

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

@robegatt 17 дней назад

And enthusiastically does so !

@geofflethbridge8732 13 дней назад

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

@ketankothari247 2 дня назад

Beautifully explained sir 👏 👌 ❤️

@thomasherpoel3132 17 дней назад

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 18 дней назад

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 18 дней назад

God bless people like you.

@em2129 12 дней назад

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

@muhdiversity7409 18 дней назад

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

@Sanjeev... 12 дней назад

Good dsp presentation thanks

@lacyrjoao2553 12 дней назад

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

@benish0r 16 дней назад

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

@BigA1 15 дней назад

The subject material is very well presented - thank you.

@AnalogDude_ 16 дней назад

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

@user-uk5ep9hm5k 17 дней назад

It was very informative. Thanks a lot.

@MusicBent 15 дней назад

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

@hmpeter 18 дней назад

Excellent video, thank you guys! 😀

@kabandajamir9844 3 дня назад

So nice thanks sir

@jeanclaudejosephbadji5418 16 дней назад

Just amazing!

@jeanclaudejosephbadji5418 12 дней назад

Thank you very much

@robertpeters9438 15 дней назад

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

@stotzing 18 дней назад

Dakujem za velmi uzitocne video

@expertis307 13 дней назад

Excelente !!!!

@youtubeaccount931 16 дней назад

Fantastic

@landspide 17 дней назад

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 17 дней назад

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 17 дней назад

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

@xxportalxx. 11 дней назад

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 10 дней назад

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. 10 дней назад

@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.

@rabie-et8zg 12 дней назад

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

@danboschen9613 11 дней назад

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).

@fas9600 17 дней назад

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

@biswajit681 17 дней назад

Magnetics design for SMPS please

@25.muhammadhilmihaidary60 17 дней назад

wow

@chinametal 11 дней назад

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 11 дней назад

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

@robegatt 17 дней назад

Optimized version would not pass coefficients everytime.

@biiber 15 дней назад

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.