When we record digital audio, the computer helpfully shows a picture of what it looks like. But most musicians are more interested in music than waveforms, so if it vaguely looks “alright” we tend to concentrate on getting the song in tune and in time with the right lyrics.
We tend to find out most of the technical stuff by trial and error. For instance our ears tell us pretty quickly that if we record music too loud it distorts and sounds horrible. The waveform meanwhile starts to look like this:
If you look closely, instead of the usual spiky peaks, some waves are squared-off at the top and bottom. Engineers call it ‘clipping’. Any fool with a computer soon learns that you have to pull down the fader a bit so that the loud bits don’t sound distorted.
Back in the days of analogue there was an equally obvious reason not to record too quietly either: your recordings got drowned out by tape hiss. So the skill lay in getting your music onto the tape as loud as possible without it actually distorting. so as to drown out the background noise.
But I fondly imagined that with digital – now tape hiss was a thing of the past – then keeping the levels high didn’t matter any more. I started recording my demos at nice low levels so as to avoid any possibility of clipping. As a result my waveforms looked like this:
What does it matter how loud you record when it’s all nice clean 16 bit digital audio, I used to say smugly. Until one day an audio engineer finally explained in mind-numbing detail what 16-bit actually means.
Almost all of it went over my head apart from this one vital fact: a digital audio file “describes” soundwaves in a series of, erm, bits. So a 16-bit audio file describes the dynamic range from silence to full volume with sixteen bits. And you can draw those bits as a series of lines like this:
It then became clear that my recordings were actually only using half of the available bits : the top eight were describing nothing at all. So in fact I was recording in low quality 8 bit audio which is, as we know, A Bad Thing.
Even if you’ve recorded your audio at too low a level, it’s still worth pushing it up afterwards as high as it’ll safely go, using the “Normalize” function in your editing software. Here’s that same audio file after being normalised:
Normalising doesn’t improve the quality of your original recording, but it will improve the results of any further processing such as converting to Mp3 or adding EQ/compression etc. It’ll also increase the output level of your sound file.
Which is all a long way of saying:
1) The higher level you record at, the higher your sound quality
2) Normalising the recording afterwards will give you better quality, louder Mp3’s