Aliasing is one of those terms that represent quite the complex concept.
So let’s dive into it.
The fact is that, all man made systems have their limits. For example, your digital audio workstation can only handle frequencies up to a certain limit.
Therefore, in digital audio, aliasing happens when a device processes data which goes beyond the device’s restriction. In digital audio aliasing happens because audio signals have a wider spectral content than the digital audio workstation can work with.
To avoid this, extra frequency components have to be removed and this is one of the roles of your audio interface.
So then, what does anti aliasing do in audio?
Anti aliasing in audio is the process of removing higher spectral content from a signal in order to ensure that the digital audio workstation can handle it and not have frequencies aliasing back into the spectral content. Which means you can sample your signal more accurately. An anti aliasing filter is what is utilized to achieve this.
Aliasing is a result of a sampling error…. therefore it is sometimes defined as the misidentification of a signal frequency, which can introduce distortion or other artifacts into the recording.
Aliasing is easy enough to avoid, but it pays to understand the basics.
The Nyquist frequency
The Nyquist theorem basically states that the highest recordable frequency is equal to half of the sample rate.
With this definition we can do some easy math.
We can basically state that at a sample rate of 44Khz we can record up to 22Khz which is simply the half or 44khz… The nyquist frequency in this example is therefore 22khz.
With that said, if a signal has any frequencies higher than the Nyquist frequency, they are interpreted by the converter and assigned to frequencies lower than the Nyquist frequency.
From here, you end up with aliasing.
To therefore prevent aliasing, you have to ensure that you record with an adequate sample rate.
You should choose the sample rate that best represents the Nyquist frequency you wish to record up to.
The default sample rate is 44.1 kHz, which means that can safely record frequencies all the way up to 22,050 Hz. This is more than enough headroom to avoid aliasing, though you can choose to record at 48 kHz for a bit more safety while also keeping file size relatively low.
Your audio interface comes equipped with anti aliasing filters that allow you to avoid aliasing.
Let’s look at how they work in principle.
When we take an analogue signal and record it into our DAW, the signal is reconstructed digitally.
Sometimes the digital reconstruction is not an exact copy of the analogue one, and we end up with errors.
An analogue-to-digital onverter will allow you to convert analogue signal into digital for recording in your DAW.
In simple terms when you record a voice, the voice is an analog that gets captured to be converted into a digital signal that you can process in your DAW.
A digital-to-analogue converter allows you to convert digital information like a recorded into analogue format which you heat come out of your speakers.
All of this is done automatically by your audio interface.
Some things that cause aliasing
Pretty much any altering done to an audio signal will make it prone to aliasing below are some processes that add spectral content to a signal.
Compressors are used to control the dynamic range of audio signals. Dynamic range is simply the volume.
It’s impossible to record a vocal without the performer being loud at some instances and being quiet at others.
Dynamic range control therfore ensures that the volume of the vocal or instrument is consistent.
The use of a compressor alters the audio signal and therefore introduces additional spectral content to the signal.
Limiters are just a type of compressor that is often times used to tame a mix by getting it as loud as possible without clipping which is essentially going over 0db.
Limiters just like compressors also add spectral content to audio signals therefore making them prone to aliasing.
Saturators are digital tools that are used to give an audio signal warmth. They do this by adding harmonic content to an audio signal.
Most Saturators mimic the sound of recording with tape saturators.
They also alter audio signals and add spectral content.
Anti aliasing filters are used in analog to digital converters, they are by design meant to filter all components of the signal that are above the Nyquist frequency in order to prevent aliasing from occurring at the digital sampling stage.
Ideally an anti-aliasing filter will be a brick-wall filter that allows everything below the Nyquist frequency to pass while filtering everything over the Nyquist frequency.
An anti-aliasing filter is important and pretty much needed whenever analog signals are sampled, or when a digital signal’s sample rate is converted from a high sample rate to a lower sampling rate.
In analog to digital converters this is a two part process.
The initial stage involves sampling at a very high rate, at which point an analog anti-aliasing filter is used.
The material is then down-sampled to the desired rate (such as 44.1 kHz, for example) where a digital anti-aliasing filter is used to achieve the final result.