Digital Audio Essay

1062 words - 4 pages

• Introduction
Technology has paved a way for us to appreciate music in a more personal and convenient ways. Gone are the days that we need a huge investment, wide space and a stationary action for us to appreciate quality music: the CDs and walkmans that render us immobile, the lack of good earphone technology, and even the low-quality amplifiers have been gone.
A more interesting part in the technology that incorporates sound and music, however, is on how studios and computers facilitate recording and compressing sound without compromising quality. Indeed, without the use of compression technology, sound recordings will take up much bigger spaces than the compact files that we enjoy today. It is to be recalled that what was once a 12-15 track CD is now conveniently able to store hundreds of tracks due to compaction. Moreover, without sound recording, there is no way that we will be able to hear tracks of our favorite bands without attending to their concerts.
An understanding of hearing, sound and its conversion into digital data is essential to properly utilize sound recording and playback facilities. It is also helpful to know the essential properties of digital audio in order to maximize sound editing techniques.
• Digital Sampling, Filtering and Compression
a. Sound Production and Digital Sampling
Sound is produced through variations in air pressure. Stringed instruments, mainly guitars, use vibrations to produce sounds. Electronic devices use loud speakers to facilitate the production of sounds (Marshall, 2011). Our ears are sensitive to the changes in pressure, thus, we perceive them as sounds.
Computers, however, cannot process analog sounds. There is a need to convert the sounds into discrete, mathematical data (Marshall, 2011). Through digital sampling, analog signals are measured in regular, equal intervals. The values of signals at the intervals are then recorded and fed into the computer through special hardware.
Digital Sampling should follow Nyquist’s theorem, where the sampling frequency should be at least twice the highest frequency component in the audio signal (Marshall, 2011). Significant loss of data would be experienced if there is a low sampling frequency. This is observable in CDs, where the maximum frequency in sampling is 44.1 kHz since the maximum comprehendible frequency is at 20 kHz.
Nyquist’s Theorem, however, has some implications. Before sampling the sound, it should be first fed into a low-pass filter to avoid phantom noises in the higher frequency regions (Marshall, 2011).
b. Audio Filtering
Composite signals, as shown in Figure 1, make up most, if not all, of the sounds we hear. These signals overlap in order to create the desired sound. Filtering adds or subtracts from these signals in order to create a synthetically-created desired sounds, much like the equalizers we use (Marshall, 2011). High and low-pass filters also demonstrate the principle by subtracting low and high frequency signals,...

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