The Analog to Digital Conversion Process
Hi, my name is Gian Faye and I'm from the Philippines.
I put up this presentation for the course
Introduction to Music Production
by Loudon Stearns
(Berklee College of Music) courtesy of
Coursera.org
.
*USE SPACEBAR OR CLICK THE ARROWS ON THE RIGHT* HAPPY LEARNING! :)
Quick Review
Sound is pressure variations in air. When sound pressure is picked up by an input transducer, sound pressure values will be transmitted as voltage. This is what we call the 'analog' of the sound.
On the other hand, computers only understand one language - binary, which is mainly composed of zeroes and ones: 1 representing ON or a positive value & 0 representing OFF or a negative value.
The Analog to Digital conversion is the process of converting the series of pressure signals, ANALOG, to clusters of 1s and 0s, DIGITAL, to be understood by the computer.
This process is also called the SAMPLING process.
Binary
A bit (short for binary digit) is the basic unit of binary information which contains only two values: 1 and 0.
To build up enough binary information, we make a collection of bits using this formula:
2^1 = 2 values (1 and 0)
2^2 = 4 values (00, 01, 10, 11)
2^3 = 8 values (001, 010, 011, 100, 101, 110, 111)
2^4 = 16 values (...) and so on and so forth.
The numbers in green are the nth power,
this is what we call the wordlength.
Wordlength
So, what does the wordlength have to do with this process? Well, let's take a CD for example. The compact disc's standard bit length is 16-bit (which is 2^16 = 65,536).
The wordlength affects the amplitude. The greater the wordlength, the larger the dynamic range is.
That is why it is recommended to record at 24-bit on your studio. This gives you a larger dynamic range, which means you can perform a wider measure of amplitude and far less likely that you could hit the distortion level.
Sampling Rate
Sampling rate is the measurement of vibrations per second.
The Nyquist theorem states that a wave must be sampled twice to get a true representation of the wave. This is because the wave has both positive and negative peaks.
So if your sampling rate is 44,100 Hz, the highest frequency possible is 22,050 Hz. If it is 48kHz,
your highest frequency is 24kHz.
Therefore, the higher the sampling rate, the higher the frequency you could reach.
Increasing the sample rate on a record speeds up the tempo of your track, decreases its time, and increases the frequency.
Decreasing the sample rate slows down and lengthens the record and also decreases its highest possible frequency.
TAKE NOTE! The standard sample rate is 48kHz, where its highest frequency (24kHz) is appropriate to what the human hearing could accomodate, and 24-bit wordlength.
Hope you learned something helpful!
If you have any suggestions, additional insights,
and cool ideas that could improve
Thank you and have a nice day.