Tutorials

Audio: Using Digital equipment – a primer on the technology.

Since so much of editing and recording is now moving into the digital domain it is important to have some idea of exactly what this means and how it will affect you. Although many community radio stations around the world still use the old fashioned ¼ inch tape for recording and editing (even the BBC World Service still makes use of it), there is an increasing trend towards originating and finalising material in a digital format.

There are some things that you need to be aware of however if you decide to go this route. It is now possible to buy digital microphones (Sony), digital recorders (Minidisc and DAT – Digital AudioTape) and digital editing software (Pro-Tools, Sound Forge, Cool Edit Pro etc).

It is even possible nowadays to broadcast a station from a computer without ever having to go into the studio – Internet radio. But, to use an old computer expression – Garbage In, Garbage Out (GIGO) – if the quality of the material is not good enough to start with, then, no matter how good your equipment is, it will still sound bad at the end.

So, what do you need to know? We’ll start at the basics and work our way up:

What is a sound wave?

When you hear a sound, you are really sensing changes in the air pressure around your eardrum. (You can think of air pressure as how densely packed together air molecules are.) However, we do not hear air pressure changes caused by the weather. Instead, we hear air pressure differences, which vary rapidly over time.

How do we perceive pitch and loudness?

Note that when there is no sound wave, the air pressure is constant. This is perceived as silence. When the sound wave reaches your eardrum, the air pressure changes above and below the normal atmospheric pressure. You perceive the amount of change as the loudness of the sound. The rate at which the air pressure is changing is perceived as the pitch. In scientific terms, loudness and pitch correspond to the sound wave’s amplitude and frequency, respectively. The frequency is usually measured in Hertz, or cycles per second. The loudness of a sound is usually measured as a fraction of a standard loudness, often in decibels (dB).

What is timbre?

The characteristic sound of a waveform is called its timbre. Timbre, also referred to as tone colour, is said to be rich or full when there are many different frequencies in a sound. Most people would consider a sound from a sine wave to be dull since it only has one frequency.

A sound’s different frequencies, each with varying amplitudes, are also referred to as the spectral content of a waveform. The spectral content, which you might say is the more scientific term for timbre, usually varies over time. Otherwise, the sound remains static and is again considered dull. The spectral characteristic of a waveform over time is the signature of a tone which allows you to refer to it as string-like or horn-like.

What is sampling?

When you put a microphone up in the air and scream, the microphone converts the changes in sound pressure into changes in electrical voltage. If you were to graph the changing voltage inside a microphone cord, it would look exactly like the graph of the air pressure going up and down. This is what is called an analogue signal.

Up until not too long ago, sound was always recorded as an analogue signal in magnetic tape or vinyl grooves. One of the problems with storing a signal in this form is that it is hard to accurately record the signal without adding noise, and when you make copies of your recording, you again have to convert to an electrical analogue signal and re-record it. Listen to a third generation cassette recording, and you’ll know what we’re talking about. Also, editing with tape is a not an easy task since you must always be fast-forwarding or rewinding to a section, splicing, etc. This is called linear editing.

With recent advances in computer technology, it has become efficient and economical to record sound waves by digital sampling. In digital sampling, the voltage analogue of the sound wave is divided and stored as numbers representing the amplitude of the wave over very small segments of time.

What is a sampling rate?

The number of times the waveform is sampled per second is the sampling rate. As you can imagine, with higher sampling rates you store more information about the sound’s changing amplitude. This gives you more fidelity. As a matter of fact, it is theoretically impossible to accurately record frequencies above one-half of the sampling rate. This threshold frequency is called the Nyquist frequency, and it should be considered when selecting a sampling rate. Frequencies higher than the Nyquist frequency show up as alias noise. The downside to very high sampling rates is that since each sample takes up space in memory, higher sampling rates will fill up your hard drive faster than lower sampling rates.

What is quantization noise?

Another factor determining the recording fidelity is how you decide to represent each sample. For example, if you were to represent the amplitude of each sample as a number from one to four, you would have to be rounding all the time to the closest value. This rounding error is called quantization noise. With 8-bit binary numbers, up to 256 different sample values are available. With 16-bit numbers, 65,536 different values are possible. Even though 16-bit samples take up twice as much space as 8-bit samples, it is recommended that when at all possible you use 16-bit samples.

What are the advantages to digital audio editing?

Once you’ve recorded a sound as a digital sample in your hard drive, you have the ability to perform edits like copying, cutting, and pasting without losing any fidelity whatsoever. As some people like to say, this is with an accuracy of up to 0.000023 seconds (single sample spacing at 44100kHz sampling rate). With a visual editors you can actually see a representation of the sound waveform to navigate through the sample quickly and accurately.

Another advantage of storing sound in the digital domain is the availability of digital signal processing (DSP) techniques, which can be used to alter the sound such as filtering, simulating room acoustics, and other special effects.

Using equipment to record, edit and play material

Recording stage:

The equipment used at this stage would consist of some form of microphone and a recording medium. In the Analogue domain this would usually be a tape machine – both quarter inch and compact cassette. For recording on tape you need to have a fairly heavy quarter inch machine, these are bulky, weigh a lot and have a battery life of only an hour or so. But they have been used in radio since the invention of tape.

Having such a wide surface on which to record the material these tapes can be of very high quality. In music studios if tape based recording is used then a one or two inch tape was the format of choice for the richness and quality of sound recorded on them.

The quarter in tape is a perfectly fine medium for recording interviews and, if necessary, more information rich pieces e.g. music.

For plain interviews it is useful to do the actual recording on compact cassette. The main reasons for this are that the recording machine is likely to be lighter, have a longer battery life and, whereas a portable quarter inch machine will only be able to record 15-30 minutes of material on one tape spool, compact cassette routinely have 90 minutes of space.

Cassette recording does need a high quality recording machine however. Normal, consumer machine will not have the high quality recording heads required for radio interview (although, in emergencies they do get used). There are a number of professional cassette machines available on the market (Sony Pro, Marantz) which are both mono and stereo. The price of these can be quite high however, especially when compared to the price/quality of digital recording equipment like the MiniDisc.

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