Hello musicians and friends! My name is Cosima and this is my second assignment for Intro to Music Production online at Coursera.org. For this assignment I’ve chosen to discuss the analog to digital conversion process. I spent some time reading up on the process and enjoyed learning something new. I hope my post will spark some interest in this topic for my readers. Thanks for visiting my blog and reading my post.
Analog to digital conversion process
In my last post I indicated that the source of an audio signal in my studio generally is a voice. The sound of that voice affects the air and creates longitudinal pressure variations that are picked up by a microphone which converts those variations into voltage variations know as an analog signal. That’s great for live performance but if we want to send that signal into a computer’s digital audio workstation (DAW) we’ll need to convert the analog wave signal to a digital signal or data.
The only thing the computer can deal with is strings of numbers. Things represented in 1 and 0s, called binary information. So, there’s a process to go from the continually variable sound into a stream of ones and zeros and that process is called a sampling process. An analog signal is a wave form that is a continuous stream of data that the computer can’t recognize whereas digital data is discrete or individually separate and distinct. To convert the analog wave into digital data of ones and zeros I’ll need to use the Analog to Digital converter in an audio interface device.
The audio interface uses a common method that converts analog to digital that involves three steps: Sampling, Quantization and Encoding.
The analog signal is sampled at an interval rate making many, many measurements per second. Most important factor in sampling is the rate at which the analog signal is sampled. Over 40,000 times per second to be able to accurately represent the continuously variable signals in the air as a digital representation. And the higher the sampling rate the higher frequency that can be represented accurately in the digital domain. And this frequency is known as the Nyquist frequency, just half a sampling rate. So a sampling rate of 44,100 hertz can accurately represent half of that in the digital domain, 22,050 hertz. The human ear can hear a range of about 20,000 hertz and the CD standard sampling rate is 44,100 hertz which will accurately represent everything we hear as human beings.
Sampling yields a discrete or individually separate and distinct form of continuous analog signal. Every discrete pattern shows the amplitude, the extent of a vibration or oscillation, of the analog signal at that instance. The quantization is done between the maximum amplitude value and the minimum amplitude value. Quantization is approximation of the instantaneous analog value.
In encoding, each approximated value is then converted into binary format of 1s and 0s the computer can then recognize which we now can manipulate in our DAW for the purpose of music production.
Thanks again for taking the time to read my post! Please feel free to leave comments. Your input is appreciated.
(sources include http://www.tutorialspoint.com and wikipedia)