How to analyse and create a filter in MATLAB for recorded audio?

Hi, I have recorded the signal in an audio file using MATLAB. How do I normalise amplitude of recorded signal to a maximum value of 1 and why does it need to be normalised?

Answer 1

[Walter Roberson]

new_signal = old_signal ./ max(old_signal(:));

However, keep in mind that it is possible that you happened to have a sample that was more negative than the maximum signal was positive, such as if the signal peaked at -2.1343 in the negative but at +2.1315 in the positive. Normalizing to maximum value 1 could result in values below -1.

"why does it need to be normalised"

Does it need to be normalized? See https://www.mathworks.com/help/matlab/ref/filter.html#bt_vs4t-2_1 the first example in filter(), which clearly shows inputs larger than 1.0

[Walter Roberson]

Note: when you use audioread(), the value is already normalized to -1 <= x <= 1 unless you specifically used the 'native' option.

[Leonardo Wayne]

Hi Walter. Thanks alot for your answer. Actually, I will need to do 4 DSP lab tutorials before attempting to do this task in full. It needs more in depth knowledge of Fourier Transform in MATLAB, usage of FDA tool and how to reconstruct the original audio signal. So I will create 4 new posts and attempt these labs before doing this task. Question: if I want to redirect you to these 4 new posts, how do I do that?

[Walter Roberson]

You could post links here. However, I did not study much signal processing so I might not be able to answer the questions.