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Yes, my form teacher doesn't understand also how is happening, because using a standard PC also introduce noise, and the PC is grounded too, it uses typical clover shaped plug, and the laboratory is grounded, so, yes, pretty strange how we're getting noise XD

Hello, so, just an update, the system is not working as intended yet, but I'm discovering some interesting things. First of all, I want to know if there's something that I should have in mind when assembling the teensy and audio boards because when I've been testing with the oscilloscope the...

Yes, the concept is the same, I've testing with some precalculated coefficients in MatLab using FX-LMS algorithm and seems kind of working, is not clear. Due to the deadline I had, I can't do more now, so the new deadline is for January, I'll check the library trick that Paul mention previously...

Hi AntiLoop, no problem with the thread, is here for learning 😊 I can´t remember properly, but I'm pretty sure that cardioid microphones mostly are dynamic microphones, they use electromagnetism to generate electric pulses that represent the sound that the membrane, which is connected to a...

Sorry, this is a mistake, d(k) is the signal measured at the point where I want the cancellation to be done, so it's the input from the second Line In, because it is measure with the microphone. The rest of terms are properly label, y(k) is the output and x(k) is the input signal to treat with...

Thanks for your answer. Regarding to option 1, I've seen that there was a queue block, but having to analyze three different points of the flow per channel, which means six in total, I think it's going to be a big problem in terms of latencies because I have to control a lot of things. On the...

Oh, I mean, if I use the FIR filter example, all is working, if I use the FIR filter as I intended, now is also working, but my new issue is how am I supposed to convert the input and output signal to numbers in order to recalculate the filter impulse coefficients in real time. In my actual...

Ok, update, I've found the issue. If I use any FFT the sound is destroyed, so I have to find a way to convert the analog input signals to digital, and operate them to get the coefficients of the fir filters, I thought the fft1024 blocks would be the way, but no, I guess the level of demand to...

Thank you for this information MarkT, it has been very useful, but right now I'm very lost with this, I have spent all week trying to get it to work in different ways and it does nothing. This is what I was thinking that will work, but even making the mixers and amps just 1.0 to be a...

Update: I change firFilter_Left.begin(FIR_PASSTHRU, numCoeffs); firFilter_Right.begin(FIR_PASSTHRU, numCoeffs); Just to directly pass the input to output as the description says, but is still not working, I'm very confused right now.

Good morning. First, I've learn one big issue, is obvious, but I didn't realize until now, I should define the array as int16_t to be properly used in the FIR filter module. So I comment the begin() from the loop and initialize this in setup(): // Debugging int16_t FIR_Coef_Deb[numCoeffs] =...

Ok, that's right. I need to update the coefficients in real time because they have to adapt to the changes in the reference signal that I'm trying to cancel, so that's why I want to change the filter. I think there's an example of FIR in the Arduino IDE, I'll check it to make something similar...

Well, to keep continuity with the theme, I don't know if I'm using the FIR filter wrong, but there is no sound at all, it's like a dam for the sound. I've been thinking about using part of the idea of the convolution code I mentioned here to do the signal calculations based on the Least Mean...

So, short answer: praying The long one, I think the latency isn't going to be the biggest problem because the point where the cancellation should take place will be enough further to give some room to processing all. The idea is to demonstrate that is possible to build something functional...

So, the easiest way to approach this would be to use a "good" external amplifier, I mean, anything better than the boards I'm finding around. For example, I could connect the microphones to the computer, route to outputs and connect it with a simple line cable with 3.5mm to 3.5mm male jack...

I'm sorry, I know you are trying to help me in the best way possible and I don't mean to hinder you. I will try to be as clear as possible about what I want to do. To begin with I think it is important to clarify the objective I am looking for. My idea is to build an active noise cancellation...

I really didn't answer to this, yes, in practical is like having 4 separate microphones, or 2 stereo (that is the same), but everything points to the use of amplifiers.

No exactly, first of all, thank you very much for your quick response Paul. I use the Line In because the idea is to have the best sample possible, so how is an electronic subsystem is not strictly necessary to use directly connected microphones, you just need the channels and you are supposed...

Hello. I'm working on my final degree thesis, and I'm trying to make an electronic subsystem capable of active noise cancellation. To achieve it, I'm using two Audio Shield (Rev. D) boards and a Teensy 4.0 connected in Quad Channel mode and I've now two line in inputs and, obviously the line...