New hardware and PCB design course on mixed-signal embedded systems just released! ⏵Course content: www.phils-lab.net/courses ⏵Course sign-up: phils-lab-shop.fedevel.education
Thank you for your work! Also interested in the IIR Filter. FWIW I googled the FMAC and found this informational video: st-onlinetraining.s3.amazonaws.com/STM32G4-Peripheral-Filter_Math_Accelerator_%28FMAC%29/index.html
Another great video Phil, thanks for clearing up how to actually obtain the filter coefficients, so far I have just trusted the calculators .. It makes complete sense that it would be the inverse FFT of the filters frequency response. It would have been nice to see the effects of scalloping prior to using the window function, just to really clarify what it does to a signal :) -Keep 'em coming, love me some DSP.
Took my undergrad DSP final 4 days ago, this video would have been incredible to have for my earlier days in that class. To all students taking DSP, they are lucky to have your incredible videos, Phil
Fantastic demonstration of FIR filters! The real-time visualization using the serial oscilloscope made the concept incredibly intuitive and engaging. Thank you, Phil, for such a clear and practical explanation!
I just discovered this video now. Smooth as butter. It is like a good action movie, well constructed, fluent, interesting and also very informative. Thanks for the great effort.
I'm in the last 6 weeks of MSc Digital Systems Engineering at the University of York, they have provided _LITERALLY_ zero DSP content... So thanks to Phil & his lab for actually providing this video.
I just thought about exactly the same!!!!!! Clear problem description, a bit of theory and an easy to understand implementation which makes the solution crystal clear, very very well done
You have done in 30 minutes what my brain has struggled with for hours - trying to bridge the gap between the practical FIR filtering code that I've examined with the theoretical/maths in Lyons book. This is one of the best tutorials I've seen on any subject - perfectly paced, clear and just the right level. Thanks!
@@PhilsLab One question after playing with your FIR Filter designer: what is the purpose of "Shift Samples"? Once you set the filter length (# of taps) isn't it implied that you will always shift by 1? Or is this a way to do decimation and filtering at the same time?
This is exactly the DSP video I needed to bridge the gap in my knowledge between theory and implementation. Keep these videos coming! I also like the fact that you share resources you found helpful.
Wow, this will come in so handy on the automotive application I'm working on, which will be polling something like 10 signals from various analogue sensors. Thanks for yet another excellent video!
Phil, your video libray of tutorials is absolutely amazing! Thank you for putting together these relatively short, practical, tutorials on extremely valuable information! 🙏
pure gold! I just learned at Special maths about convolution and Dirac impulse, Heavyside and so on but never though how to make the connection between theory and practice! As I said, pure gold! THANK YOU!!!!
Another amazing video. Great job! Your stm32 coverage is better than any I've seen on youtube, and you have such a wide breath of topics from initial board layout to fab, testing and implementing drivers and filters!
Just, Thank you for all your time and hard work to educate people. So refreshing to see complex concepts, explained thoroughly and implemented in a place where anyone can see the results. Merry Christmas mate.
This is amazing. I am still very new and naive to dsp, but it’s amazing to see your content. Thank you so much for your willingness to share. It will not be forgotten ❤
Very good video and teaching. Very clearly explained. Thank you! Just FYI... a good way to increment a circular buffer index is like this: circular_index = (circular_index + 1) % BUFFERLENGTH;
You are Awesome!! I have been looking at all of your videos, Just finished the ADXL355 build and took 20 pages of notes. I am trying so hard to understand all of this but it is hard for me because I am a self taught and just want to learn. You do a perfect job at teaching what you are doing.
Thank you so much! this was so helpful, specially the firt part of the video, so I could better understand the concepts. This video was uploaded only 6 months ago, I'm glad I'm taking this course now
In a previous attempt to understand FIR, I used Excel to model a Low Pass Filter. Rather than impulse, I used a step (square wave) and altered the response until it matched a sine wave. Having the proper math would probably be better, but it worked as a one-off.
Thank you once again for the amazing job you put in your videos. First digital control theory, then firmware configuration + hardware design and now this. It's like you are reading my mind or smth. Really amazing stuff. I'm all for a Q&A video!
Really great video, watched the one after it about IIR as well. This is what RUclips has been missing. Put more math. If people run away then they are your wrong audience. I loved it, I tried to show some how this was done and had difficulty with tools to bring a proper demonstration. This video will be my goto from now on.
Thank you so much! Yeah, I'm always unsure of how much maths to put in with it still being accessible to most people. I'll try and sprinkle in more and more and see when people have had enough :)
Amazing video Phil! The only thing that was getting to me was that your voice was competing with your clicky keyboard, which made it difficult to follow the software as you spoke. But that is by no means an indication of the quality content you share!!... So please keep on sharing your knowledge with us.
Where you format the accelerometer output before calling CDC_Transmit_FS(...) sprintf() returns the number of characters written to logBuf[] so you can save yourself the call to strlen() which is yet another pass over logBuf. I'll keep my comments about using sprintf() in ISR's for another day :-) ... Ok, well, this is relatively low rate stuff so formatting and transmitting from another ringbuffer in the main loop fed by the ISR would be another option to keep the ISR's lean and mean. Great stuff.
Great video man! FIR is really powerful, even my total bonehead implementation of 32 values moving average manages to increase DHT11 termometer resolution by more than order of magnitude.
I wish there was a bit of electronics demonstrations. I don't know actually where all you are talking about is in electronics world. 1. What hardware setup you are using? is it the microcontroller dev board or it's another piece of hardware? 2. What level of electronics this happens? audio, wifi, bluetooth signals...etc. ? What I liked: 1. The code explanation is very good especially following up other drivers libraries you've done in other videos 2. Theoretical information about is good even I didn't get a clear idea of where FIR should be implemented?
Even though the main topic of these videos seems to be µC programming, they still give the most understandable explanations about digital filters i ever watched. Could you perhaps a video on Fast Fourier Transform FFT and its Inverse (IFFT) implementation in a similar way of your FIR and IIR videos ? Greetings
Wow I wish I saw this video a year ago! I hope you continue this series. It would be neat if you did a video using DSP to do hand motion detection on a microcontroller.
Programming tip: If you keep buffer length a power of 2, i.e. 2^4=16, you can reset the index faster using: fir->bufIndex = ++fir->bufIndex&(FIR_FILTER_LENGTH-1)
One comment: if filter length is L=2^N, then the group delay of the output comes in-between the samples: (2^N - 1) / 2. For example, for N=3, the group delay is 3.5. This can be important, if filter data need to be combined with the input data.
Makes's me think i should think about a way to implement a FIR on my stock investments, with a large pass band :). That Said, thanks for the vidéo, great job and nicely structured.
Nicely presented. The C code can be made a lot more performant by storing two copies of the impulse response and dropping the branch in the inner loop.
Hi. Thanks for the great video. Two questions though; 1. You mentioned; anything below 10 Hz should have a DC Gain of about 1. What do you mean by DC Gain here in case of a low pass filter? 2. Before windowing, you mentioned that the filter response is not nice because at about 10 Hz, the reduction is only -6 dB or -5 dB. How much reduction should we expect for a great response after windowing? Thanks in advance.
Very cool training course, so well done, I love it. Don’t forget to setup a PayPal URL to support your activities for me and others. Here is some minor programming feedback. - for the sprintf use always the snprintf function e.g.: snprintf(logBuf, sizeof(logBuf), …) This is a good programming practise to avoid buffer overflows. - For circular buffers the wrapping can be made easy via modulo operations. e.g.: fir->buf[fir->firbufIndex++ % FIR_FILTER_LENGTH] = inp; This way you never need to check if your are at the end of a buffer or at the beginning) Again really great stuff you are doing here. PS: I would write this as a C++ class where the constructor gets the length of the filter as a parameter and allocates the float array, however for the audience here keeping it simpler with basic C is better because everyone must understand the individual steps. Regards from Hannover, Helmut
Two last days I've being reading articles about PCM DSD, and now you have published this video. And this is not a random vid but rather one of my subscriptions. I wonder what does it mean ))) And how do you have time for all these cool things for us?
Amazing tutorial and very clean code. When i handle interrupts, what I do is set a flag when the interrupt is identified, then immediately clear the interrupt. Then that then triggers execution of the appropriate callback in the super loop. I do this to keep the interrupt vector free and clear. What do you think about this approach? Of course if I was using FreeRTOS, I would have some kind of thread safe semaphore or even a queue. But damn man your explanation of FIR filter was spot on and probably the best I have ever seen. Edit: also, subscribed! Keep up the amazing work!
First, I found topic and the information very useful. Thank you! However, please consider a nice headset microphone. (A gaming headset blocks external noises nicely.) At times, the keyboard clicking exceeded the volume of the narration. It really broke my chain of thought and was quite painful. Not sure why, but around 19 min into the video the keyboard sounds seem to get extremely loud.
Would the time lag not be from the impulse response being shifted right, and then your t = 0 filter tap would be the middle tap and the fir-> out would always be from in this case 8 samples earlier as it is 16 taps shifted by 8? Maybe I am misunderstanding how this works
Thanks, really great content. This is a legit intro to dsp. One thing I would suggest to help improve the clarity of your content is looking into typesetting math with LaTeX (or whatever people are using nowadays). I'm sure you will continue to have to present math and that really helps make the symbols and arrangement clear.
Thank you, Jon! I love Latex and use it for many more 'academic' things. However, I prefer the simplicity of handwritten notes for quick videos like this.
Hi Phil, thanks for the content.i m not and STM32 expert, Do you think the fpu unit can help speeding up calculations? Does it have fft implemented in hardware ?
Your Nyquist limit says your rate must be less than fs/2. If it were equal to fs/2 the output woudl be a costand related to the phase of you sampling with fs/2. And if your signal is just a fraction under fs/2 your filter will take infinite time or samples to show the signal. So that is a good lead-in to what is a good sampling rate compared to Nyquist and how does that effect time response?
It would be interesting to understand whether replacing the x[n] * h[n] convolution with the FFT product X(f)•H(f) and then returning back to y[n] with IFFT could be advantageous or not.
Thank you. The real question is why the sensor generates noise in the first place? When it is static there is no noise, but when you swing it we see a lot of added noise that shouldn't be there I think. If the noise is part of the true data it should be left.
Thank you for this video. It's helpful. One question I have is at ~4:30, it's said that the output length is "in general" the same as the input length. For FIR, how can this be? If I take a 4-sample sequence and convolve it with a 4-sample filter kernel, the output is (2*4-1) 7 samples long.
New hardware and PCB design course on mixed-signal embedded systems just released!
⏵Course content: www.phils-lab.net/courses
⏵Course sign-up: phils-lab-shop.fedevel.education
Do you have any plans to look at implementing IIR and FIR filters using the STM32G series FMAC?
Thank you for your work! Also interested in the IIR Filter.
FWIW I googled the FMAC and found this informational video: st-onlinetraining.s3.amazonaws.com/STM32G4-Peripheral-Filter_Math_Accelerator_%28FMAC%29/index.html
Another great video Phil, thanks for clearing up how to actually obtain the filter coefficients, so far I have just trusted the calculators .. It makes complete sense that it would be the inverse FFT of the filters frequency response. It would have been nice to see the effects of scalloping prior to using the window function, just to really clarify what it does to a signal :)
-Keep 'em coming, love me some DSP.
Your vid was totally intelligibly explained. IMO the depth of the theory was on point 👍🏻. Also, i am looking forward to hear how IIR differs from FIR
I would also love some more videos on control theory! Starting at PID, which you have, and continuing to LQR and Kalman.
Took my undergrad DSP final 4 days ago, this video would have been incredible to have for my earlier days in that class.
To all students taking DSP, they are lucky to have your incredible videos, Phil
Thank you so much for your very kind comment, George! Hope all went well with your DSP final :)
Fantastic demonstration of FIR filters! The real-time visualization using the serial oscilloscope made the concept incredibly intuitive and engaging. Thank you, Phil, for such a clear and practical explanation!
I just discovered this video now. Smooth as butter. It is like a good action movie, well constructed, fluent, interesting and also very informative. Thanks for the great effort.
Thank you very much for your kind comment!
This class is increadible, so clear. Wow just wow.
I'm in the last 6 weeks of MSc Digital Systems Engineering at the University of York, they have provided _LITERALLY_ zero DSP content... So thanks to Phil & his lab for actually providing this video.
These videos are much clearer and more concise than many university courses and textbooks. Well done Phil!
I just thought about exactly the same!!!!!! Clear problem description, a bit of theory and an easy to understand implementation which makes the solution crystal clear, very very well done
You have done in 30 minutes what my brain has struggled with for hours - trying to bridge the gap between the practical FIR filtering code that I've examined with the theoretical/maths in Lyons book. This is one of the best tutorials I've seen on any subject - perfectly paced, clear and just the right level. Thanks!
Thank you very much, Robert!
@@PhilsLab One question after playing with your FIR Filter designer: what is the purpose of "Shift Samples"? Once you set the filter length (# of taps) isn't it implied that you will always shift by 1? Or is this a way to do decimation and filtering at the same time?
This is exactly the DSP video I needed to bridge the gap in my knowledge between theory and implementation. Keep these videos coming! I also like the fact that you share resources you found helpful.
Thank you Paul, you've made this complex topic so much more approachable.
this is the best video I have watched to explain FIR filters so far. thank you
Wow, this will come in so handy on the automotive application I'm working on, which will be polling something like 10 signals from various analogue sensors. Thanks for yet another excellent video!
Phil, your video libray of tutorials is absolutely amazing! Thank you for putting together these relatively short, practical, tutorials on extremely valuable information! 🙏
Thank you very much! I'm vey glad to hear your comments on these videos.
pure gold! I just learned at Special maths about convolution and Dirac impulse, Heavyside and so on but never though how to make the connection between theory and practice! As I said, pure gold! THANK YOU!!!!
That's great to hear, thank you very much, Lucian!
Thanks for this quick FIR refresher and for mentioning t-filter. I found it to be extremely easy to understand and utilize for my STM32 projects.
Really really precious tutorial!
The git repository is also a treasure as well!
Great job, Phi!
Cheers Phil! You make these topics very digestible and encouraging!
Thank you very much, very glad to hear that! :)
Another amazing video. Great job! Your stm32 coverage is better than any I've seen on youtube, and you have such a wide breath of topics from initial board layout to fab, testing and implementing drivers and filters!
Just, Thank you for all your time and hard work to educate people. So refreshing to see complex concepts, explained thoroughly and implemented in a place where anyone can see the results. Merry Christmas mate.
Thank you so much for those very clear explanation ! Your voice is perfect for non English speaker !
Thank you very much!
Thank you very much for these two superb an clear videos. Please keep teaching us about digital filters with this “hands-on” aproach.
Cheers!!
Thank you so much, Alberto! Many more videos to come :)
This is extremely useful as I'm currently working on an application based on imu data and require signals with very little noise. Thanks a ton !!
That's great to hear, thank you. Hope this can help out with your project!
This is amazing. I am still very new and naive to dsp, but it’s amazing to see your content. Thank you so much for your willingness to share. It will not be forgotten ❤
Thank you very much :)
Very good video and teaching. Very clearly explained. Thank you!
Just FYI... a good way to increment a circular buffer index is like this:
circular_index = (circular_index + 1) % BUFFERLENGTH;
You are Awesome!! I have been looking at all of your videos, Just finished the ADXL355 build and took 20 pages of notes. I am trying so hard to understand all of this but it is hard for me because I am a self taught and just want to learn. You do a perfect job at teaching what you are doing.
Thank you! Really glad to hear that - many more videos to come :)
Thank you so much! this was so helpful, specially the firt part of the video, so I could better understand the concepts. This video was uploaded only 6 months ago, I'm glad I'm taking this course now
Thank you very much for watching!
Amazing video and as an Eclipse contributor it’s nice to see other people using Eclipse for C/C++ as well :)
I'm stunned! Thanks for the class! Absolutly flawless! Keep up the great work!
Thank you very much!
Thanks for this - you've put a lot of effort into your video! Appreciated.
Thank you so much for the content you create. These videos are immensely helpful and I appreciate the time and effort you put into this.
In a previous attempt to understand FIR, I used Excel to model a Low Pass Filter. Rather than impulse, I used a step (square wave) and altered the response until it matched a sine wave.
Having the proper math would probably be better, but it worked as a one-off.
Thank you once again for the amazing job you put in your videos. First digital control theory, then firmware configuration + hardware design and now this. It's like you are reading my mind or smth. Really amazing stuff. I'm all for a Q&A video!
Awesome, I'm glad you like them, Patrick! Let's see if I can read your mind with the next video :D
really clear much more clear than my teacher used to explain this!
Really great video, watched the one after it about IIR as well. This is what RUclips has been missing. Put more math. If people run away then they are your wrong audience. I loved it, I tried to show some how this was done and had difficulty with tools to bring a proper demonstration. This video will be my goto from now on.
Thank you so much! Yeah, I'm always unsure of how much maths to put in with it still being accessible to most people. I'll try and sprinkle in more and more and see when people have had enough :)
Excellent. Looking forward to your new course.
12 hours of lectures in DSP < this video. Thank you for doing this, it has been incredibly helpful!
Very glad to hear that, thank you for watching!
A superb job in demonstrating from concept and theory to product.
Thanks for another awesome video! 😁 I'm love watching your channel while working on my projects 😇
Awesome, thank you very much - very glad to hear that! :)
I just wana thank you Phill to sharing your knowledge with us!
Appreciated that.
Thank you very much, Tomáš!
Excellent explaination and excellent keboard sound.
Amazing video Phil! The only thing that was getting to me was that your voice was competing with your clicky keyboard, which made it difficult to follow the software as you spoke. But that is by no means an indication of the quality content you share!!... So please keep on sharing your knowledge with us.
Thanks!
Kalman filter in discrete time is pretty interesting. May be you can describe in details Madgwick filter?
Also very interested in that! Cheers
Also interested. Cheers
Where you format the accelerometer output before calling CDC_Transmit_FS(...) sprintf() returns the number of characters written to logBuf[] so you can save yourself the call to strlen() which is yet another pass over logBuf. I'll keep my comments about using sprintf() in ISR's for another day :-) ... Ok, well, this is relatively low rate stuff so formatting and transmitting from another ringbuffer in the main loop fed by the ISR would be another option to keep the ISR's lean and mean. Great stuff.
Great video man! FIR is really powerful, even my total bonehead implementation of 32 values moving average manages to increase DHT11 termometer resolution by more than order of magnitude.
This is awesome! Really hope for some more DSP videos!
Thank you, next video is on Z-transforms!
Great work as usual Phil! Keep up it mate 👍
Thank you very much!
Your channel is a treasure. Thank you
Thank you very much!
Thank you, useful content again. Additionally, I suggest SerialPlot Real Time Plotting Software to plot your data.
Hi Phil great video. Very informative
I wish there was a bit of electronics demonstrations. I don't know actually where all you are talking about is in electronics world.
1. What hardware setup you are using?
is it the microcontroller dev board or it's another piece of hardware?
2. What level of electronics this happens? audio, wifi, bluetooth signals...etc. ?
What I liked:
1. The code explanation is very good especially following up other drivers libraries you've done in other videos
2. Theoretical information about is good even I didn't get a clear idea of where FIR should be implemented?
I repeat what others said: WOW. GREAT STUFF!!
Thank you very much, Sanel!
very good video. Summarizes a lot of hours of explanation by my "signal processing" professor :D Thanks!
Thanks, Joey!
This is really gorgeous. Thank you so much. God bless you.
And I'm gonna rewrite them using fixed point arithmetics to make life happier for low end MCUs ! go on mate !
Even though the main topic of these videos seems to be µC programming, they still give the most understandable explanations about digital filters i ever watched. Could you perhaps a video on Fast Fourier Transform FFT and its Inverse (IFFT) implementation in a similar way of your FIR and IIR videos ? Greetings
This channel is a goldmine!
I’ve been looking for such good explanations for a long time!! Thanks!
It was amazing to see this video. Keep up the good work. Please make more of such videos. :)
WOW PAUL! I am so happy with this one! You are such a good boy!
Welp, I better pick myself up some stationery then! Superb video :)
Thank you. Very instructive. I would've perform the convolution using the Discrete FFT.
amazing video phil
another good educational video, good work phil really enjoyed this.
Thank you, very glad to hear that!
Excellent work man. Thank you for sharing your knowledge in such a good way.
Thank you very much!
Very well explained, thank you. Waiting the next videos...
Thank you very much!
Fantastic! Cant wait for the Kalman one!
Wow I wish I saw this video a year ago! I hope you continue this series. It would be neat if you did a video using DSP to do hand motion detection on a microcontroller.
Thank you! I'll be making a couple more DSP videos. Next ones on IIR filtering and Extended Kalman Filters.
Thank you very much, Phil!
Thank you for watching, Rodrigo!
Programming tip: If you keep buffer length a power of 2, i.e. 2^4=16, you can reset the index faster using: fir->bufIndex = ++fir->bufIndex&(FIR_FILTER_LENGTH-1)
One comment: if filter length is L=2^N, then the group delay of the output comes in-between the samples: (2^N - 1) / 2. For example, for N=3, the group delay is 3.5.
This can be important, if filter data need to be combined with the input data.
Great video, perhaps would be nice to see a fixed point Implementation, or maybe compare the two on uC with or without a FPU.
This is what I needed desperately. Skill upgraded :)
Awesome, glad to hear that, Frank!
Nice content Phil, it helped me out a lot
Clearer than sun.
Thanks you.
Awesome, thanks!
I've learned so much from your videos. You're awesome! Thank you!
Makes's me think i should think about a way to implement a FIR on my stock investments, with a large pass band :).
That Said, thanks for the vidéo, great job and nicely structured.
Been done already.
Nicely presented. The C code can be made a lot more performant by storing two copies of the impulse response and dropping the branch in the inner loop.
Hi. Thanks for the great video. Two questions though;
1. You mentioned; anything below 10 Hz should have a DC Gain of about 1. What do you mean by DC Gain here in case of a low pass filter?
2. Before windowing, you mentioned that the filter response is not nice because at about 10 Hz, the reduction is only -6 dB or -5 dB. How much reduction should we expect for a great response after windowing?
Thanks in advance.
Very cool training course, so well done, I love it. Don’t forget to setup a PayPal URL to support your activities for me and others.
Here is some minor programming feedback.
- for the sprintf use always the snprintf function e.g.: snprintf(logBuf, sizeof(logBuf), …)
This is a good programming practise to avoid buffer overflows.
- For circular buffers the wrapping can be made easy via modulo operations.
e.g.: fir->buf[fir->firbufIndex++ % FIR_FILTER_LENGTH] = inp;
This way you never need to check if your are at the end of a buffer or at the beginning)
Again really great stuff you are doing here.
PS: I would write this as a C++ class where the constructor gets the length of the filter as a parameter and allocates the float array, however for the audience here keeping it simpler with basic C is better because everyone must understand the individual steps.
Regards from Hannover, Helmut
Two last days I've being reading articles about PCM DSD, and now you have published this video. And this is not a random vid but rather one of my subscriptions. I wonder what does it mean ))) And how do you have time for all these cool things for us?
Haha maybe I can read your mind :D It's getting a bit tough with time but I'm trying my best to get two vids out a month :)
Your keyboard is very passionate.
Awesome, needed this ❤️
Thank you, glad to hear that!
Great video! Question: What options are available to minimize the time delay between the raw and filtered signal?
Amazing tutorial and very clean code. When i handle interrupts, what I do is set a flag when the interrupt is identified, then immediately clear the interrupt. Then that then triggers execution of the appropriate callback in the super loop. I do this to keep the interrupt vector free and clear. What do you think about this approach? Of course if I was using FreeRTOS, I would have some kind of thread safe semaphore or even a queue.
But damn man your explanation of FIR filter was spot on and probably the best I have ever seen.
Edit: also, subscribed! Keep up the amazing work!
really great video sir!!
First, I found topic and the information very useful. Thank you!
However, please consider a nice headset microphone. (A gaming headset blocks external noises nicely.) At times, the keyboard clicking exceeded the volume of the narration. It really broke my chain of thought and was quite painful. Not sure why, but around 19 min into the video the keyboard sounds seem to get extremely loud.
Excelente video. Muchas gracias Phil's
Thank you for watching, Wilson!
Would the time lag not be from the impulse response being shifted right, and then your t = 0 filter tap would be the middle tap and the fir-> out would always be from in this case 8 samples earlier as it is 16 taps shifted by 8? Maybe I am misunderstanding how this works
Thanks, really great content. This is a legit intro to dsp.
One thing I would suggest to help improve the clarity of your content is looking into typesetting math with LaTeX (or whatever people are using nowadays). I'm sure you will continue to have to present math and that really helps make the symbols and arrangement clear.
Thank you, Jon!
I love Latex and use it for many more 'academic' things. However, I prefer the simplicity of handwritten notes for quick videos like this.
Thanks for sharing this awesome thing.
Really really great video, thank you for this. Great job
Thank you very much!
Hi Phil, thanks for the content.i m not and STM32 expert, Do you think the fpu unit can help speeding up calculations? Does it have fft implemented in hardware ?
As always love your work ❤️❤️
10:12 Do you think this truncation window length is what Chord refers to as "taps number" in their dacs?
Are you using a keyboard with Cherry MX Blue switches by any change?
Your Nyquist limit says your rate must be less than fs/2. If it were equal to fs/2 the output woudl be a costand related to the phase of you sampling with fs/2. And if your signal is just a fraction under fs/2 your filter will take infinite time or samples to show the signal. So that is a good lead-in to what is a good sampling rate compared to Nyquist and how does that effect time response?
It would be interesting to understand whether replacing the x[n] * h[n] convolution with the FFT product X(f)•H(f) and then returning back to y[n] with IFFT could be advantageous or not.
Amazing channel 👏👏👌
Thank you. The real question is why the sensor generates noise in the first place? When it is static there is no noise, but when you swing it we see a lot of added noise that shouldn't be there I think. If the noise is part of the true data it should be left.
great video and great job !!!
Thank you for this video. It's helpful. One question I have is at ~4:30, it's said that the output length is "in general" the same as the input length. For FIR, how can this be? If I take a 4-sample sequence and convolve it with a 4-sample filter kernel, the output is (2*4-1) 7 samples long.