The real world that we live in has an infinite variation of sound and light. This infinite variation is referred to as analog. For the sound to be used in a computer it must be digitised.
The elements covered in this summary are:
This material addresses aspects of the following syllabus outcomes:
H1.1 applies and explains an understanding of
the nature and function of information technologies to a specific practical
H2.2 explains and justifies the way in which information systems relate to information processes in a specific context
h2.1 analyses and describes a system in terms of the information processes involved.
H2.2 develops and explains solutions for an identified need which addresses all of the information processes
H3.1 evaluates and discusses the effect of information systems on the individual, society and the environment
H3.2 demonstrates and explains ethical practice in the use of information systems, technologies and processes
H4.1 proposes and justifies ways in which information systems will meet emerging needs
H5.2 assesses the ethical implications of selecting and using specific resources and tools, recommends and justifies the choices
H7.1 implements and explains effective management techniques
H7.2 uses methods to thoroughly document the development of individual and team projects.
Source: Board of studies NSW, Stage 6 Information Processes and Technology, Preliminary and HSC Courses (2007)
Whether the sound comes from an old LP, cassette tape or recording of the natural world all sound that is not generated electronically is analog.
The first step in the digitisation of any audio is to determine how the audio will be used. For example, a digital audio recording for Internet streaming has very different requirements to a recording for an audio CD. When analog files are saved to a computer there are many different formats that the file can be saved as along with many different settings. Each file format has a slightly different purpose. The .wav and .aif file types are uncompressed digital audio and are best suited for creating Audio CDs or master files while audio formats such as .mp3, .mpg, .mpeg, .rm, and .wma, are best for computer and internet presentations because of their compression.
To convert analog sound to a digital format we require an analog-to-digital converter (ADC). This function is normally built into a sound card.
The method used to digitise sound is called sampling. When a sound file is sampled we take a slice of sound at discrete intervals. The more often this slice is recorded the better the quality. Imagine you are in an experiment and have been asked to listen to a conversation. In this experiment you can only listen fortwo seconds at a time before the sound is cut off. The more often the sound is turned on the more of the conversation you pick up. In fact if we were able to turn the sound back on really quickly you would miss very little of the conversation. This is the way sampling works and the number of slices taken during a period of time is called the sample rate. The higher the sample rate the better the quality of the sound being recorded. So the larger the sample rate the better the quality because more data has been retainedand the larger the file size. Lets have a look at a table with sample rates.
|Sample Rate (the larger the number the more often the sound is sample in a fixed timeframe)||Quality and Use|
|11 kHz||low sample rate but good for speech|
|22 kHz||satisfactory music playback|
|32 kHz||broadcast quality|
|44 kHz||Commercial audio CD quality|
|48 kHz||digital audio tape|
can be seen from the illustration on the left that data is lost during the
process of digitisation. It is unavoidable. However the loss is so small that it
is negligable and often the final product can be enhanced anyway.
image courtesy of bettscomputers.com
Although many file formats are used to store digitised sound, music, and even instrument or synthesizer tracks on computers in reality the choices become more limited. Limitations occur as a function of the output device being used. For example most DVD and CD players will not accommodate most of the sound formats.
Other factors that need to be considered include sound quality, the types of equipment used for playback, and trade-offs between file sizes (smaller files mean more music) and listening quality (more compression normally means lower audio fidelity).
Some compression methods are "lossy," which means that data is lost during compression. This is a one-way, irreversible process. Other compression methods are "lossless", which means that no data is lost during compression and an uncompressed version of data can be generated from the compressed form. This is a two-way, reversible process. Lossy compression schemes usually result in smaller files.
Another key ingredient in the audio compression process is a special piece of software called a codec, short for the type of "compression/decompression" algorithm that's used to translate uncompressed data into a compressed form and to decompress for access or playback. Codec Central is a great source of information on codecs of all kinds.
table 1 provides a list of many common digital audio file formats and some details about them
|Advanced Audio Coding||aac||1997||Lossy||Yes||Also known as MPEG-4 AAC. Supports various compression rates, commonly used for iPods.|
|Audio Interchange File Format||aiff||1999||Lossless||No||Original, uncompressed Mac sound format. Not cross platform and not used in devices. format is not supported by all web browsers.|
|Compact Disk Digital Audio (CD-DA)||cda||1979||Lossless||No||Standard digital audio CD format, uncompressed but not used in many devices except CD players.|
|Musical Instrument Digital Interface||.mid or .midi||1982||lossless||no but not required||The MIDI (Musical Instrument Digital Interface) is a format for
communicating music information between electronic sound devices like
synthesizers and PC sound cards. MIDI format is flexible and can be used
for basic to professional music.
However NOTE The downside of MIDI is that it cannot record sounds (only notes). MIDI files do not contain sampled sound, but a set of digital musical instructions (musical notes) that can be interpreted by a PC's sound card. The best part of the MIDI format is that MIDI files can be extremely small. For example a 5 minute file might only be around 23K.
|MPEG-1 Level 3 (MP3)
Moving Pictures Experts Group
|.mp3 or .mpga||1993||Lossy||Yes||Uses perceptual coding to compress signals at various compression
MPEG format was originally developed for video by the Moving Pictures Experts Group. MP3 files are the sound part of the MPEG video format. MP3 is very popular format. MP3s combine good compression rates (small files) with high quality.
|OGG Vorbis||ogg||2000||Lossy||Yes||Ogg Vorbis is used to store and play digital music in much the same way as MP3, VQF, AAC does at present. It is different from these other formats because it is completely free, open, and unpatented. It has been designed to completely replace all proprietary, patented audio formats. (supported on some devices).|
|QuickTime Audio||mov||1997||Lossy||Yes||Same MPEG-4 technology as AAC, but limited usability because not used in devices.|
|Real Audio Media||rma||1995||Lossy||Yes||Proprietary, compressed audio format but limited usability because not used in devices. The RealAudio format was developed for the Internet by Real Media. The format also supports video including streaming.|
|Waveform sound files||wav||1991||Lossless+||No||Used for system sounds on Windows PCs (not used for high quality music or in devices).|
|Windows Media Audio||wma||1997||Lossy+||Yes||Offers various rates of compression or no compression (supported on most devices).|