Sample rate and bit depth is the core of what makes up the quality in an audio recording. Without proper knowledge of the theory behind these terms, it is very easy to overlook them as simply another option in the start up of your sequencer. This blog will attempt to unpack the theory and facts behind these terms and numbers in the simplest way possible. This blog may not be full of fun ideas for your music production, but it is extremely important to have a secure knowledge of the subject of how your audio works.
The way a computer processes any data is a combination of 1’s and 0’s called “binary code”. This data comes together to form a language that allows programs to perform specific tasks. When you choose a sample rate of 44.1Khz, you are telling you computer to write a piece of data at 44100 times per second, called a sample, which is the substance of what makes up your audio files. As you increase your sample rate (48kHz, 88.2kHz, 96kHz and 192kHz) it is essentially improving the resolution of the audio, writing a greater number of samples per second as you increase your sample rate.
It is important to bear in mind that increasing your sample rate will exponentially increase the file size of your audio, so make sure you have large hard drives! The other issues involved is higher sample rates will consume more of your computers CPU power, and also lower the amount of tracks or “voices” available in your session.
Compact discs (CD’s) playback at a sample rate of 44.1kHz and 16-bit, begging the question as to why we might require a higher sample rate/bit depth to record audio for music. Here is where is gets a little nerdy, so pay close attention!
The “Nyquist-Shannon Sampling Theorem” states that ‘perfect reconstruction of a signal is possible when the sampling frequency is greater than twice the maximum frequency of the signal being sampled’.
Healthy human hearing ranges from 20Hz to 20kHz, so therefore according to the ‘Nyquist Theorem’, if we want to produce a signal (a sound) of 20kHz, we must be running a sample rate (44.1kHz) that is DOUBLE the frequency that we want to record, this is to avoid an error called ‘aliasing’.
Aliasing refers to an effect that causes different signals to become indistinguishable when sampled.
So, back to why we need higher sample rates. A lot of musical instruments produce frequencies and overtones called “harmonics” far above 20kHz and out of the range of human hearing. If we want to accurately record these frequencies, we need to be running a sample rate that is DOUBLE the frequency that we wish to record.
Yet still, more questions; if we cannot hear the overtones at 40kHz (we would need a sample rate of 80kHz for this), why should bother recording at a higher sample rate?
These ultra high harmonics produced by the instrument also affect the lower tones that we can hear, for example 40kHz affects 20kHz, 10kHz, 5kHz, 2.5kHz (see the pattern?) and so on.
Recording at higher sample rate of 88.2kHz will enable you to recorded higher harmonics, giving your recordings more depth, clarity, crispness and sparkle. 88.2kHz is also double a CD’s sample rate of 44.1kHz, so when it comes time to crunch the numbers down, your computer is simply halving the number of samples used to make up your audio – a much simpler mathematical equation to process then coming down from 48kHz or 96kHz.
Beyond all the technical mumbo jumbo and numbers, (let’s face it, we are musicians not mathematicians) what really matters is how it sounds and feels. Some well-trained ears can pick the difference between a sample at 48kHz and 96kHz, however this doesn’t mean that you cannot make quality-sounding recordings using 44.1kHz from the start of your project.
At the end of the day, chances are that if someone is going to listen to your music, it is likely to be played back on a portable device, at a low bit-depth MP3, through some tiny single driver ear phones, making a lot of this sample rate business irrelevant… doesn’t that just warm your heart?