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# The Analog to Digital Conversion Process

Assignment for week 2 of Introduction to Music Production
by

## Sergio Sánchez

on 13 February 2014

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#### Transcript of The Analog to Digital Conversion Process

Computers use binary digits called
bits
(0s or 1s) to quantify each sample that is taken.
The number of bits used to define a value is referred to as the binary word length or bit depth
, and is equal to 2 to the nth power( ) where "n" is the number of bits in the binary word.
The Analog to Digital Conversion Process
Analog audio signals
must be digitized before they can be manipulated by any DAW as
Digital Audio.
Digital Audio is a representation of an Analog audio signal
, which can be stored, copied, read and manipulated without losing quality.
The process where Analog audio signals are translated into digital information is known as
Analog to Digital Conversion
, also called
A/D Conversion
. Two essential factors that affect the A/D process are:
Sample Rate
and
Bit Depth.
Sampling is the process of
taking discrete readings of a signal at various moments in time.
it's like taking a
snapshot
of the signal in a particular instant and when the samples are played back in succession approximate the original signal.
The
sample rate
is the frequency with which these snapshots are collected. The sample rate for digital audio is determined by the
Nyquist Theorem
.
The Nyquist Theorem states that:
in order to produce an accurate representation of a given frequency of sound, the sampling frequency should be at least twice the highest frequency contained in the signal.
For example, consider a signal composed of a single sinewave at frequency of
1 Hz.
If we sample this waveform at
2 Hz
(twice the frequency of 1 Hz)
is sufficient to capture each peak and trough of the signal.
If we sample at a frequency higher, for example
3 Hz
, then there are
more than enough samples to capture variations in the signal.
But if we sample at a
frequency lower
than 2 Hz, for example 1.5 Hz, there are
not enough samples to capture the peaks and troughs in the signal,
then sampling will produce wrong tones, known in digital audio as
alias tones.
Because range of human hearing goes from
20 Hz to 20 kHz
, this law indicates that a
sampling rate of at least 40 kHz is required to capture full frequency audio
. that's why we use
44.1 kHz
and
48 kHz
as a standard sample rate for cd's and dvd's.
An
A/D Converter
must be able to accurately represent differences in amplitude. The relative
amplitude of a sample is captured through a process known as quantization
. This simply means that
each sample is assigned to the closest available amplitude value.
As such,
larger binary words are able to quantify variations in amplitude with much greater accuracy.
The
dynamic range
capability of an
A/D system
can be calculated by multiplying the word size by six, for example:
A system with a
16 bit binary word
would produce a
dynamic range of 96 dB.
16 x 6 = 96
while a
24 bit system
would have a
dynamic range of 144 dB.
24 x 6 = 144
A/D
Interface
for example:
A/D
Interface
Pro Tools 101 Official Courseware, Version 9.0.
http://redwood.berkeley.edu/bruno/npb261/aliasing.pdf
Hi! my name is
Sergio Sánchez,
i'm from Mexico, and this is my presentation for week 2 of Introduction to Music Production at Coursera.org. This time my assignment is from "
The Analog to Digital Conversion Process
".
Sample Rate
Bit Depth
The
Analog to Digital

Conversion
is a complex but interesting process, it's important to understand the concepts of
Sample Rate
and
Bit Depth
so we can figure out what's better for our session and project, I hope this review has been helpful, thank you for your feedback, good luck and good vibes.
Just remember:
larger binary words required higher hard drive storage space
.
Full transcript