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Analog to Digital: The Conversion Process

Isabella Barbato – May 2016

+Let me first introduce myself

My name is Isabella, and I’m originally from

Italy

I’ve been living in Singapore for the past 7 years, and I love it!

New to music production, but LOVE

music! Currently studying music theory

& guitar

In this presentation I’ll talk about the Analog to Digital Conversion

Process

This is my submission for Week 2 of

“Introduction to Music Production”,

an online course offered by Berklee

University for Coursera students

+Some definitions

n  Sound = pressure variations in the air, continually variable

n  ADC = analog-to-digital conversion; all about turning audio signals into strings of 1’s and 0’s – information that can be interpreted by a computer

n  Bit = A bit (short for binary digit) is the smallest unit of data in a computer. A bit has a single binary value, either 0 or 1

+How does analog-to-digital conversion work?

n  Audio analogue to digital conversion works by measuring the amplitude (volume) of an incoming electrical pressure sound wave (an electrical voltage), and outputting these measurements as a long list of binary bytes.

n  In this way, a mathematical “picture” of the shape of the wave is created

+Two essential parameters

n  Remember join-the-dot pictures? To produce a good image you must have sufficient dots to capture the detail of the shape & the dots must be positioned accurately. Quality in a join-the-dots picture depends on:

n  Number of dots

n  Accuracy of the positioning of the dots

n  In the same way, there are 2 important parameters which control the quality of the audio conversion process:

n  Sample rate (number of measurements of amplitude per second)

n  Word length (accuracy of each measurement of amplitude)

+Sample Rate

+What is sample rate?

n  Sample rate is simply the number of samples (or measurements of amplitude) taken per second.

n  Sample rate is also sometimes known as: n  Sample frequency: e.g. CD quality sample

rate is expressed as "44.1KHz", meaning simply that the converter takes 44,100 measurements of amplitude per second. Sample frequency is independent of the frequency of the sound waves being converted.

n  Sample bandwidth

+Sample rate is constant n  Once set, sample rate does not vary during a recording, although

different audio files recorded at different sample rates may be used together in a multi-track system if the software permits it

n  Usually, as in the case of a DAW, audio files of differing sample rates will need to conform (be converted too) a single sample rate, typically 44.1KHz, 48KHz, 96KHz or 192KHz. This sample rate is usually set in the application preferences for the recording session

n  It’s good practice to set the sample rate as slightly higher than a CD sample rate (i.e. 44.1KHz) = recommended to set at 48KHz

REMEMBER! n  Higher sample rates produce better quality recordings but also

bigger file sizes which demand greater space on storage devices (such as hard drives), and faster processors (CPUs) to manipulate.

n  Lower sample rates produce poorer quality but also smaller file sizes which demand less of storage systems, CPUs and will transfer over networks (internet) faster.

+Example of sample rates

Format Sample rate

Audio CD 44,100 samples per second (44.1KHz)

DVD Up to 96,000 (96KHz)

Professional multi-track recording 48KHz or 96KHz, and even sometimes 192KHz

MP3 Variety of sample rates. The trade off is always between quality and file size

+Nyquist theory

During his research into digital audio, Harry Nyquist (a scientist) produced a simple rule to determine appropriate sample rates for differing sounds.

"The sample rate should be a little over twice the amount of the highest audio frequency (harmonic) to be recorded if poor sound quality is to be avoided".

As humans can hear audio frequencies as high as 20KHz (20,000cps/Hz), a minimum sample rate of 44.1KHz (or 44,100 sample measurements / sec) was decided upon:

n  Human audio spectrum = 20Hz to 20,000Hz (20KHz) ... therefore ...

n  Highest audio frequency = 20,000Hz ... therefore ...

n  20,000 x 2 = 40,000 + "a little bit more" = 44,100 samples per second 

At the time, 44.1KHz was considered the best compromise of quality and file size. Over tome, there has been much debate over the importance of using higher samples rates. However, given the the quality of much modern domestic audio equipment produces increasingly inferior sound, it is unlikely that many consumers will benefit anyway

NOTE: Increasing the sample rate above 44.1KHz does not dramatically improve the sound. Increasing word length (see later) has a greater impact

+Word Length

+Word length

n  Word length expresses the number of bits in a byte.

n  For example, a byte made up of 8 bits (e.g. "10011101") has a word length of 8. We can also say the byte is 8-bit. In simple terms, the longer the word length the more detailed or precise the information can be, so a 16-bit byte can contain more "precise" or "detailed" information than an 8-bit byte.

n  Word length determines the number of possible variations a byte can be. For example, there are 16 possible variations in a 4-bit byte (see chart on the right)

+Word length & quality

n  In audio files, higher word length means better sound quality

n  In audio quality terms, more accurate measurements mean less distortion of the true shape of the sound wave

n  CD quality is 44.1KHz / 16-bit. This means that every second a converter will produce 44,100 16 bit numbers

n  It is recommended, when setting up your DAW, to set 24-bit as standard word length, even though the finished mixes must be converted to 16-bit prior to audio CD duplication

+Recording at too low a level

n  It is important to recognise that even if a converter has a high word length, setting the record level too low will result in a smaller range of bits being used and effectively reduce the word length of the recording

n  Setting the record level too high however, will risk digital clipping, an unpleasant distortion that is the result of all sample measurements that exceed the upper limit of the word length range of the system being quantised down to the highest available value. For example, in a 16-bit system this might be 1111111111111111

n  It is therefore important that recordings are made at the highest possible level without clipping, which explains why the signal is often passed through an audio limiter before it enters the A to D converter

+Additional Reading & References:

n  [Video] Analog to Digital Conversion Explained for Audio - Sound Engineering (YouTube)

n  Digital to analog conversion and sound (University of Texas)

n  What is an analog to digital converter? (Planet of Tunes)

n  Nyquist-Shannon Sampling Theorem (Wikipedia)

n  Nyquist Frequency (Wikipedia)

n  Analog to digital conversion (Slideshare)