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# The Analog to Digital conversion process.(Week 2 to Music Production - Coursera.org)

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## Cursos de Sonido CFP

on 19 March 2013

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#### Transcript of The Analog to Digital conversion process.(Week 2 to Music Production - Coursera.org)

Hi! I´m Silvia, from Argentina

This lesson is for week 2 of Introduction To Music Production at Coursera.org. In this prezi, we´ll be learning about
The Analog to Digital conversion process. In this lesson ,
we´ll talk about: 1 So first of all, Analog and digital 2 So, how does it work? How do we convert one into the other? Analog to digital process 3 Digital sound parameters Sample rate / Resolution so that´s all
for now .. Silv .- - Analog and digital (definition)

- Analog to digital
conversion proccess

- Digital audio properties
(sample rate / resolution)

-DAW: how to set
sample rate and resolution Definition So, as we´ve seen in week 1,
sound is pressure variations in the air continually variable.
It's not discrete steps,
but constant. and that is Whenever a mic. converts that signal into voltage
(that goes trough a cable, an amplifier or stereo) the electrical current vary similarly to the variation in the air pressure generated by the sound wave. It is also continuous,
constant, like pressure variations. That´s why it is called analogue (similar) air
presure = current
voltage + - Positive /negative
voltage positve/negative
presure + - In this lesson,
we´ll talk about Variations on current voltage are similar
to the air presure variations of a sound wave
(they are both continuos) compression/rarefaction Digital systems, on the other hand, need to convert audio signal into data that can be processed by numerical calculations. In fact, it has to convert binary information 0 1 0 1 0 1 0 1 0 1 0 0 0 1 0 1 0 1 0 1 0 1 0 0 1 1 1 0 1 0 1 0 1 0 1 1 1 0 1 0 1 0 1 0 1 1 0 1 0 1 0 1 0 0 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 streams of 1 and 0 analog signal voltage variation into + - Binary information is based primarily on the bit,

a single kind of memory location
And a single bit is a 1 or a 0. So, if we have a single bit,
we can only represent two things: 1 bit = 1 or 0 But, If we want to represent larger numbers, we have to start a collection of bits, collecting bits into "words" two numbers
(1 or 0), ON or OFF heads or tails
from a coin... 01011 011
0001
101010111 000111 1 0 1 0 1 11 10 1110101
010101 1111110 In music production we use diferet lengh words. For example: MIDI = 7 bits words

Digital Audio = 16 bits word
So, an specific number of bits (word length), alows you to represent an specific number of values posible
values 1 bit
word 2 bits
word 00, 01, 10, 11 = 4 =2 0, 1 posible
values 3 bits
word =8 posible
values 000, 001, 010 , 011
100, 101, 110, 111 and so on... 0 1 01 01 0101 0101 0 0101
0 1 01 01 0101 0101 0 0101

010 010101 0 01010 010 010101 0 01010
010 010101 0 01010 0 01010
010 010101 0 01010
010 010101 0 01010
010 010101 01 1101 0101011 1110 010
1010 111 010101 1010 111 010101
1010 111 010101 1010 111 010101 Every audio interface, has an Analog-Digital Converter (A / D),
which is responsible of processing the audio analog signal,
in order to get a binary representation of the sound the amplitude level is measured up in short periods of time, dividing the signal into samples, and assigning a binary value for each sample´s amplitud This process is called,
Analog to digital Conversion,
or Sampling So, when an analog audio signal is picked up by the A/D
in order to convert it into digital, 1100101011111010 0101110101011111 0111010100110111 each sample, gets a binary number,
(a collection of bits or "word")
that represents its amplitude There´s one important detail to keep in mind: It stopped being a continuous
(acoustic wave / analog signal) to become a discrete signal,
which is discontinuous,
separated by steps. With this process, the audio signal
has changed a fundamental characteristic 01 01010101 0 0101 10
00 101010 010 01 001010
010 000 10101 010 0 10101 11 01010
010 010110 111 And this is an important thing you should know, because as we gain something with digital audio, we also loose something else. - Digital Audio offers high precision, simplified editing and processing, it can be storage in smaller spaces, and will not loose audio quality over time -But Analog sound is known to provide more sound depth and warmth, though is prone to produce higher noise levels, and to deteriorate within time. So, as we said earlier, in the analog to digital process, the A / D converter will begin to split the signal into samples separated by identical time intervals and will measure each sample´s amplitud,
in order to represente it with a binary number The amount of samples measured per seconds,
is called Sample Rate
(or sampling frecuency) It is measured in Hz, and it will determine the highest frequencies we can record
(both fundamental frequencies or those from the spectrum) the higher the sampling rate,
the higher frequency that can be represented
accurately in the digital domain The highest frequency that can be represented accurately in digital domain is known as the Nyquist frequency The higer frecuency that can be recorded,
will be half your sampling rate. since you need a minimun of 2 samples
to represent 1 period of the wave So, in order to record a given frequency,
we need twice the sampling rate and it depends on the sample rate That explains the CD standard 16bits | 44.100Hz resolution sample rate Since the sample rate of 44.100Hz, provides the necessary means to represent the highest frequency that humans can hear (22.000Hz) Sample
Rate quality Resolution On the other hand,
the amount of bits (word length) used
to represent the amplitud of each sample is called Resolution
(or bit depth) It is measured in bits and it will determine
our dynamic range Bit depth (word length), is related to amplitude a higher word lenght, results in more values to
messure the amplitud of each sample,
leading to a higher dynamic range in other words, it will guarantee
a lower noise/ higher signal amplitud recording,
leading to a wider dinamic range In other words, we´ll get more depth: a bigger dynamic range between the softest sounds and highest One last tip:
Anyway, In the studio, you should use
a higher resolution (word length). If posible, use
24-bit (or 32bits float)
This will give you is a wider dynamic range,
and it will aloud you to record a little quieter
-avoiding clipping-
and yet, get a very good recording,
(since the noice gets lower) As we know, the CD standard resolution is 16 bit So, when recording, use higer sample rate and resolution. It will lead to a more accurate digital representation of the sound you are recording DAW settings Finally, So here, I leave you a few graphichs
that show how to set this up in:

Ableton Live