I rewrote to this: %Duration and Tone %sawTone = @(l,t) repmat(linspace(-1, 1, t*FS), 1, round(l/t)); sawTone = @(d,t) repmat(linspace(-1, 1, t*FS), 1, round(d/t));

DSP theory says a linear time-invariant system can be perfectly modelled by convolution with its impulse response. An actual spring reverb unit is probably not perfectly LTI, but close enough that convolution gives a reasonably convincing impression of it. Since the spring reverb impulse response is just an audio signal, what I'm doing in this video is to try and reconstruct it from a model of some of its components that I analyze with the plots I show in this video.

The x axis there is just the index into the array. The first value gets plotted at 1, the second value at 2, and so forth. Nothing fancy!

Thankq

It looks like you are asking the zeros function to make an array that has a fractional number of elements. Try using the round, floor, or ceil functions to remove the fractional part of the size of the array and turn it into an integer value instead.

Same issue here using the exact code from the video, maybe it's caused by using a newer version of Octave. As Dan recommended using a round function made it work: instead of bassTone = zeros(1, bassNoteDur * FS); use bassTone = zeros(1, round(bassNoteDur * FS));

Fuck this game. Nothing I could do.

yes!

UPDATE: I realized my mistake! For some reason copying and pasting the beats vector from the melody to the base confuses Octave because it thinks you want get different values out of the same vector, so I just had to make a new arrangement for the bass notes :) tl;dr Octave LOVES counterpoint lol

Yeah, you know all about sounding intelligent, completely naturally.