What is 44.1 kHz sampling rate?

44.1 kHz is a number that gets thrown around a lot when people talk about digital audio. But what the heck does it mean? 44.1 kHz is the “sample rate” or “sampling frequency.” It has to do with how analog sound waves, like the ones that come out of musical instruments or people’s mouths, get converted into the digital audio files that we listen to on our phones, computers, and CD players.

Analog vs Digital Audio

See, sound in the real world is “analog.” That means the sound waves are continuous – they go up and down smoothly. But computers and digital audio devices can’t directly store or understand analog signals. They need the sound to be “digital,” – meaning the smooth analog wave gets chopped up into a bunch of individual numbers that represent the height of the wave at each moment in time.

Samples

Each of those chopped-up moments is called a “sample.” How many of those samples are captured per second is what we call the “sample rate,” which is measured in “hertz” or “Hz.” One hertz just means one sample per second.

So when we say an audio file has a sample rate of 44.1 kHz, that “kHz” stands for “kilohertz” – aka “thousands of hertz.” 44.1 kHz means that the audio comprises 44,100 individual samples for every second of sound. That’s a lot of samples!

Why 44.1 kHz?

44.1 kHz is widely used as the standard sample rate for digital audio. It can be found in MP3 files, WAV files, audio CDs, recording software, and the default settings of digital audio workstations or DAWs. But why 44.1? How did we settle on that particular number?

Nyquist-Shannon Sampling Theorem

It comes from the Nyquist-Shannon sampling theorem. This theory states that if you want to capture an analog signal in digital form perfectly, your sample rate needs to be at least double the highest frequency you want to record.

Most humans can hear the highest frequency, around 20,000 Hz or 20 kHz. Double that, and you get 40 kHz. The 44.1 kHz sample rate is slightly above that threshold, giving extra headroom.

The CD Standard

When audio CDs first emerged in the 1980s, the folks developing the standards wanted the digital audio to sound as close as possible to the original analog recording. They settled on 44.1 kHz as the sample rate to use on the CD format, which is a big part of why 44.1 kHz is still the most common sample rate today. Many of our digital audio ecosystems were built around the standards set by the CD.

Alternatives to 44.1 kHz

Now, 44.1 kHz isn’t the ONLY sample rate out there. There are higher sample rates, like 48 kHz, 88.2 kHz, 96 kHz, and even 192 kHz. Higher sample rates can record higher-frequency sounds and provide more detail. They’re sometimes used in professional recording and audiophile contexts.

However, 44.1 kHz is still the standard for most consumer playback systems and music and audio production. It provides excellent sound quality, is compatible with the broadest range of audio hardware and software, and results in smaller file sizes than higher sample rates.

How 44.1 kHz Works

Let’s explore in more detail how that 44.1 kHz sample rate works in practice. How does an analog signal get converted to 44.1 kHz digital audio?

Analog-to-Digital Conversion

It happens through a process called analog-to-digital conversion, or “ADC.” When an analog signal, like sound from a microphone or an electric instrument, goes into an ADC, the converter measures the signal’s voltage 44,100 times every second. The signal level at each moment is rounded to the nearest number in the digital system and recorded.

This process happens super fast, 44,100 times each second, chopping the smooth analog wave into a stepped series of 44,100 individual voltage measurements that approximate the original shape of the analog signal.

Bits and Bit Depth

Now, how precise are each of those 44,100 measurements? That depends on something called the “bit depth.” Just like the sample rate measures how many samples are taken per second, the bit depth measures how detailed each sample is.

CDs and many other digital audio use 16-bit audio, meaning there are 2^16 or 65,536 possible levels each sample can be rounded to. Higher bit depths like 24-bit audio allow for even more precise measurements and finer dynamic range.

Reconstruction on Playback

The process happens in reverse when it comes time to play back that 44.1 kHz 16-bit audio file. The series of 44,100 discrete measurements per second gets sent to a digital-to-analog converter, or “DAC.” The DAC uses those numbers to reconstruct a continuous analog voltage signal that approximates the shape of the original sound wave. This gets sent to your headphones or speakers and becomes the sound you hear. And voila! Digital audio at work.

Lossless vs. Lossy Compression

One more related concept worth mentioning is compression. Audio files can take up a lot of space, especially at high sample rates and bit depths, so many digital audio formats use compression to make the files smaller.

There are two main types of compression: lossless and lossy. Lossless compression squeezes the data efficiently into a smaller package but in a way that allows it to be perfectly reconstructed again later. Formats like FLAC and ALAC are lossless.

Lossy compression cleverly discards some data that most people can’t hear, resulting in much smaller files that sound very similar to the original. MP3 and AAC are standard lossy formats. But with a lossy file, you’re not getting those original 44,100 samples per second anymore—some are approximated or discarded for file size.