In The Studio: Digital Audio 101—The Basics

Frequency Bandwidth & Sampling Rate

Sampling rate is probably the area of greatest confusion in digital recording. The sample rate is how fast the computer is taking those “snapshots” of sound.

Most people feel that if you take faster snapshots (actually, they’re more like pulses than snapshots, but whatever), you will be capturing an image of the sound that is closer to “continuous.” And therefore more analog. And therefore more better. But this is in fact incorrect.

Remember, the digital world is capturing math, not sound. This gets a little tricky, but bear with me.

Sound is fundamentally a bunch of sine waves. All you need is at least three point values to determine a sine wave function that crosses all three. Two will still leave some ambiguity – but three – there’s only one curve that will work. As long as your sample rate is catching points fast enough you will grab enough data to recreate the sine waves during playback.

In other words, the sample rate has to be more than twice as fast as the speed of the sine wave in order to catch it. If we don’t hear more than 22 kHz, or sine waves that cycle 22,000 times a second, we only need to capture snapshots more than 44,000 times a second. Hence the common sample rate: 44.1 kHz.

But wait, you say! What if the function between those three points is not a sine wave. What if the function is some crazy looking shape and it just so happens that your A/D only caught three that made it look like a sine wave?

Well, remember that if it is some crazy function, it’s really just a further combination of sine waves. If those sine waves are within the audible realm they will be caught because the samples are being grabbed fast enough. If they are too fast for the our sample rate it’s OK, because we can’t hear them.

Remember, it’s not sound, it’s math. Once the data is in, the computer will recreate a smooth continuous curve for playback, not a really fast series of samples. It doesn’t matter if you have three points or 300 along the sine curve – it’ll still come out sounding exactly the same.

So what’s up with 88.2, 96, and 192 samples/second rates?

Well, first, it’s still somewhat shaky ground as to whether or not we truly don’t perceive sound waves that are over 22 kHz.

Secondly, our A/D uses a band-limiter at the edge of 1/2 our sampling rate. At 44.1, the A/D cuts off frequencies higher than 22 kHz. If not handled properly, this can cause a distortion called “aliasing” that effects lower frequencies.

In addition, certain software plug-ins, particularly equalizers suffer from inter-modular phase distortion (yikes) in the upper frequencies. The reason being, phase distortion is a natural side effect of equalization – it occurs at the edges of the effected bands. If you are band-limited to 22 kHz and do a high end boost, the high end brickwall stops at 22 kHz.

Instead of the phase distortion occurring gradually over the sloping edge of your band, it occurs all at once in the same place. This is a subject for another article, but ultimately this leaves a more audible “cheapening” of the sound.

Theoretically a 16-bit recording at 44.1 smpl/sec will have the same fidelity as a 24-bit recording at 192. But in practicality, you will have clearer fades, clearer reverb tails, smoother high end, and less aliasing working at higher bit depths and sample rates.

The whole digital thing can be very complicated – and in fact this is only touching the surface. Hopefully this article helped to clarify things. Now go cut some records!


Matthew Weiss is the head engineer for Studio E, located in Philadelphia. Recent credits include Ronnie Spector, Uri Caine, Royce Da 5’9” and Philadelphia Slick.

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