Sample Rate 101: The Key to High-Quality Audio

Are you confused about the concept of sample rates in audio recording? 

Setting your sample rate too low can make your tracks sound like you recorded them in the 1940s. 

Set it too high and your DAW can experience lags and crashes. 

That’s why it’s important to know the reasoning behind setting it where you do. 

Sample Rate Overview

The sample rate is the number of “snapshots” or samples a system takes of a sound wave (analog audio signal) per second. 

It’s important to note that computers can’t process and store sound waves. 

Therefore, a recording device like an audio interface converts sound waves into a digital format – a series of digital numbers that computers can understand. 

It does this through a process called sampling. Hence the name sample rate. 

The sample rate is the number of samples that a recording device takes of an analog audio signal per second. 

An audio interface uses these samples to create a digital sound wave representation. 

This process allows you to record and store audio on your computer – in its digital form. 

What to Expect From This Post

In this post, we’ll take a deeper dive into the importance of sample rate in audio production. 

We’ll discuss what sample rate is, how it’s measured, its impact on the quality of your recordings, and what sample rate you should record at.

With a deeper understanding of sampling rates, you’ll be able to make informed decisions to optimize your audio recording and mixing. 

What is Sample Rate? 

To understand sample rate, it helps to know that there’s a difference between how our ears and computers process sound. 

How Humans Process Sound

Analog sound waves from nearby noises, our voices, and instruments cause our eardrums to vibrate. 

Our inner ear contains fluid that converts these vibrations into electrical impulses (signals). 

Then, our brain interprets these signals allowing us to distinguish between different pitches, volumes, and sound qualities. 

Long story short, computers can’t do what our ears and brains do.

They process information by reading digital signals. 

How Computers Process Sound 

A computer’s central processing unit (CPU) interprets digital signals – represented in binary code (digital 1s and 0s) – to perform various tasks. 

Therefore, we need a sound card or audio interface to record audio into a computer. 

Analog to Digital Converter (ADC)

Sound cards and audio interfaces have what’s known as an analog-to-digital converter (ADC).

An ADC samples an audio signal (sound wave) and converts it into a digital representation for your computer. 

It does this by measuring a sound wave’s amplitude at certain intervals – determined by the sample rate. 

Then the ADC gives each sample a numerical value based on the audio signal’s amplitude at a given point. 

Once your ADC converts an audio signal into a digital format, you can use your digital audio workstation (DAW) to store, edit, and manipulate it.

With that said, the sample rate is the number of times an ADC measures the amplitude of an audio signal per second in hertz (Hz). 

Theoretically, a higher sample rate allows an ADC to capture more snapshots per second, resulting in a more accurate representation of the original audio signal. Or in other words, a higher audio resolution. 

Most Common Sample Rates in Audio Production

The most common sample rates in audio recording are 44.1, 48, are 96 kHz. 

Sampling rates are based on the idea that to record and recreate a frequency in the human hearing range you must be able to sample it at least twice every wave cycle. 

Since the human hearing range is between 20 Hz and 20 kHz, for the best quality we should record at a sample rate of at least 40 kHz

Most audio engineers record slightly above 40 at 44.1 or 48 kHz. 

Reasons to Record at 44.1 or 48 kHz and Not a Higher Sample Rate


44.1 kHz is the default sample rate in audio production.

 In the 1980s, 44.1kHz was established as the de facto sample rate for Compact Discs (CDs). It has since become the standard for most digital audio formats, including MP3 and WAV files. 

Why 44.1 kHz?

Because it can accurately reproduce frequencies up to 22.05 kHz – slightly above the highest frequency audible to the human ear. 

File Size 

While recording at a higher sample rate than 44.1 kHz may seem like a better option, note that higher sample rates produce larger file sizes. 


Remember how we discussed sample rate as the number of samples an ADC takes of an audio signal per second? 

At a sample rate of 96 kHz, an ADC captures 96,000 samples per second compared to 44,100 at 44.1 kHz. 

All of this information has to be stored somewhere. 

The higher you set your sample rate, the more data your computer will have to store. 

44.1 kHz offers a nice tradeoff between file size and audio quality. 

It’s important to note that although 96 kHz captures more data than 44.1, it doesn’t necessarily produce better audio quality. 

96 kHz can accurately reproduce frequencies up to 48 kHz.

Reproducing frequencies beyond 20 kHz – the upper limit of human hearing – is pointless in many cases. 


Since an ADC captures more samples per second at higher sample rates, your computer’s CPU will have to work harder to process more data. 

You may experience glitches, dropouts, or even crashes if your CPU isn’t equipped to handle an increased workload. 

Reasons to Record at 96 kHz

Many mixing and mastering engineers prefer to work with tracks at higher resolutions, particularly, 96 kHz. 

From there, they’ll downsample to 44.1 kHz to fit the audio standard. 

The reason behind recording, mixing, and mastering at higher resolutions is that some plugins create harmonics beyond the limit of human hearing (20kHz). 

For example, some compression or distortion plugins produce harmonics above 20 kHz. 

With these plugins, recording or mixing below 96 kHz can cause your software to “fold back” frequencies beyond 20 kHz so they fit into the audible frequency range – causing distortion. 

The process of folding back frequencies to fit within a lower frequency range is known as aliasing foldback

In summary, aliasing occurs when the sample rate of a recording is too low to accurately capture higher frequencies. This phenomenon causes your software to fold back the highest frequencies in your recording (even though we may not hear them) which can cause distortion. 

To reiterate, aliasing fold-back is an issue when you’re working with plugins that have an internal sample rate of 96 kHz. 

Audio professionals who work with various plugins that oversample prefer to record, mix, and master tracks at 96 kHz. 

From there, they’ll downsample to 44.1 kHz to fit the audio distribution standard. 

What Sample Rate Should You Record At? 

In most cases, you should record at 44.1 or 48 kHz

As mentioned before, 44.1 kHz is the standard sample rate for CDs and other digital audio formats. 

This rate also offers a nice balance between audio quality, CPU usage, and file size. 

Any rate above 48 kHz will put more strain on your CPU and create larger audio files while not affecting your audio quality. 

However, if you’re a mixing or mastering engineer who uses a lot of saturation or compression plugins, you may benefit from recording, mixing, and mastering at 96 kHz. 

Simply put, some plugins reproduce higher frequencies more accurately at 96 kHz. 

Recording and mixing at lower sample rates can cause aliasing which leads to distorted high frequencies. 

From there, you can convert your project to 44.1 kHz through a process known as “sample rate conversion”. 

How to Change Sample Rate in Your DAW

FL Studio

To change your sample rate in FL Studio, go to Options > Audio Settings. 

In the right corner, you’ll see “Sample Rate (Hz)”. 

Select the sample rate you’d like to work in from the dropdown menu. 

Ableton Live 

In Ableton Live, go to Preferences > Audio > Sample Rate. 

Under In/Out sample rate, select the sample rate you’d like to work in from the dropdown menu.

Logic Pro

In Logic Pro, go to File > Project Settings > Audio. 

Select the sample rate you’d like to work in from the dropdown menu. 

Changing your sample rate is similar for every other DAW. 

Takeaway: What is Sample Rate in Audio Production?

Sample rate refers to the number of times your recording device samples incoming audio signals per second.

A higher sample rate means your device will capture more samples, resulting in a more accurate representation of your recorded audio. 

However, there is a limit to how much increasing your sample rate improves accuracy. 

Any sample rate beyond 44.1 kHz will have little to no effect on your audio quality. That is unless you’re working with plugins that have an internal sample rate of 96 kHz. 

If you are, recording and mixing at a higher sample rate can provide a better representation of high-frequency content in your audio. 

However, it is important to keep in mind that recording and mixing at a higher sample rate also increases the amount of data your computer needs to process and store.

Therefore, working with higher sample rates will require more storage and processing power.

Ultimately, your ideal sample rate will depend on the plugins you use and your computer’s resources.

Generally, I recommend recording and mixing at 44.1 kHz – the standard for CDs and other audio formats.

44.1 kHz provides a nice balance between audio quality, CPU usage, and file sizes.

But, consider working at 96 kHz if you’re using plugins that oversample. Setting your sample rate higher, in this case, will prevent aliasing. 

Aliasing can cause distorted high frequencies if your session sample rate is lower than the sample rate of your plugin. 

If you have any more sample rate questions, don’t hesitate to leave them in the comments below!