I’m relatively new to the whole digital signal processing thing as I got a new module on that this semester. We already covered a bunch of filters such as Low-Pass, High-Pass, Notch, etc. Theoretically, I understand how all of these filters affect the audio signal, but now we need to design filters digitally in JUCE and I honestly don’t know how to start as we’re not allowed to use JUCE DSP’s inbuilt filters.
Let’s say, I want to create a simple first-order low pass filter plugin that is capable of handling varying cut-off frequencies (through a slider) at runtime. Referring to this video https://www.youtube.com/watch?v=uNNNj9AZisM, I’d like to design this as an FIR filter where the output is acquired by convolution using the current input and the impulse response.
I’ve seen that DSP provides functions to compute these input response coefficients, at least in IIR::Coefficients.
How could I design that filter assuming that I’m getting the raw data in the “processBlock” function in the audio processor?
Maybe it would be more efficient to design an IIR filter, so let me know that too, please.
Thanks in advance for your help!