Hi, my name is vinícius Bernucci.

I live in Brazil, Minas Gerais, Belo

Horizonte. I work with studio for over ten years. Also I make music school at the
Universidade Federal De Minas Gerais. In this work, I want to talk a little about the
nyquist theorem.

The Nyquist Theorem

According to the nyquist Theorem, in order to digitally encode the desired

frequency bandwidth, the selected sample rate must be at least twice as high as the
highest recorded frequency (sample rate 2 X highest frequency). Thus, as áudio signal
with a bandwidth of 20 KHz would require a sampling rate of at least 40.000
samples/second. In addition, it`s of equal importance that no áudio signal greater than
half the sampling frequency enter into the digitization process. If frequencies greater
than one-half the sample rate are allowed to enter into the conversion process,
erroneous frequencies - Known as alias frequencies – would enter into the audible
áudio signal band as false frequencies and produce audible harmonic distortion.

In order to eliminate the effects of aliasing, a low-pass filter is placed before

the analog-to-digital (A/D) conversion stage. In theory, a filter that would pass all
frequencies up to the nyquist cutoff frequency and have infinite attenuation thereafter
would be ideal.

In the real world, however, such a “brick wall” filter doesn`t exist. For the
reason, a slightly higher sample rate must be chosen in order account for the
attenuation slope required for the filter to be effective. A sample rate of 44.1 KHz, for
example, has been chosen in order to accurately encode an effective bandwidth up to
20 KHz.