Audio Basics : Analogue vs Digital audio


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Analogue Audio



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All sound originates from a mechanical vibration which travels through a medium as a pressure wave. The medium is typically air, however it can travel through other mediums such as water, metal, earth etc. to our ear which then produces electrical signals analogous to the vibration which our brains interpret as sound. Analogue audio is all about mechanical devices producing electrical signals when exposed to pressure waves i.e. a microphone or the reverse principle electrical signals producing mechanical movements which result in pressure waves i.e. a speaker.

For much of the 19th & 20th centuries right up until the late 1960's all audio recording and reproduction was done by analogue devices such as the Gramophone, Phonograph and Tape Recorder. Even today microphones and speakers are essentially analogue devices, primary because our larynx and ears are essentially analogue devices. It's the capturing, storage and manipulation of sound which has become digital.

Digital Audio



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Digital audio has it's roots in 2 significant technological developments, the transistor and the computer, the former because it made the valve obsolete paving the way for significant developments in analogue signal amplification & processing and the later because it revolutionised the storage, editing and reproduction of audio files.
Digital audio involves a process where by analogue electrical energy is converted in to a numerical representation by a process termed "Analogue to Digital Conversion" or ADC.

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An Analogue to Digital Converter works by sampling the amplitude of a waveform at periodic intervals and storing a numerical value in memory against a time base.

The image above illustrates a typical analogue waveform in red with the digital samples which will approximate the waveform numerically. The x axis represents time while the y axis represents the amplitude of the signal sampled. In digital audio the sample rate, expressed in Hz indicates how many times a second a sample is taken of the audio source file. The greater the sample rate the higher the quality of the reproduction. CD quality audio is sampled at 44 Khz for example while many podcasts are at 22 Khz. Most digital recorders will allow you to set this rate prior to recording. The higher the sample rate the more samples required to be stored in memory hence a larger file size. Computers store numbers in bytes of information and bytes can be either 8,16 or 32 bit. The more bits in a byte the larger the number which can be represented. In digital audio equipment and software such as a audacity you can choose the byte size 16, 24 or 32. For high quality recording and post production editing 32 is preferable however 16 is sufficient for most work especially podcasts.

The reverse process of digitising analogue audio is referred to as "digital to analogue" conversion or DAC . This takes the numerical values stored in memory and creates a analogue waveform which can then be amplified and fed to an analogue device such as a speaker.

Typically audio is converted to digital data, stored on a medium such as a hard disk, cdrom etc. where it can be manipulated by software and then converted back to analogue audio when a human needs to hear it.

In analogue systems copying files was a degenerative process (ie copies of copies became progressively worse than the original) whereas with digital systems the data file can be cloned endlessly without destroying the ordinal file. The capacity to compress audio for transfer over the internet and digital audio players such as ipod's has ushered in a revolution in how we produce, exchange and listen to audio.