Sampling Essentials – Part 1

Sampling Essentials – Part 1

10 April, 2008 | 12.00PM

• Section: Music News Topics: Guide To Synthesis, Technology
• Written by: francispreve

Now that we’ve covered the ins and outs of subtractive/analog synthesis, we’ll begin exploring other popular methods of tone generation.

This week, we’ll tackle the specifics of sampling.

Since most samplers also rely strongly on subtractive synthesis techniques, the material in the previous tutorials is equally relevant here.

If you haven’t already checked out the earlier lessons, you might want to do so before reading further.

On the other hand, if you’re already up to speed, it’s time to dive in. 

What’s a sampler?

In simplest terms, a sampler is a musical instrument that records audio, stores it digitally, then allows you to trigger the recording and/or change its pitch in real-time via MIDI using a keyboard, sequencer, drum pads, etc.

A sampler - or any digital audio recorder - converts audio to binary information via something called an analog-to-digital converter.

The a-d converter samples the voltages in an analog audio signal and converts those voltages into numerical data that the computer can store in memory or on its hard drive.

From there, the data is played back via a digital-to-analog converter and the result is a replica of the original sound.

Much of the conversion process - and overall recording quality - is governed by two variables: sampling rate and resolution.

Sampling Rate

The sampling rate is the number of times per second that the voltage is sampled.

In the same way that film records moving images by taking a series of still images and playing them back in rapid succession, a sampler records voltages at the sampling rate, then plays them back in rapid succession, thus recreating the original waveform.

As with video and film, the greater the sampling rate, the better the sample quality.

In fact, the minimum acceptable sampling rate for musical audio is 44.1 kHz.

That’s 44,100 times per second.

44.1 kHz may sound extremely fast, but that’s because the highest frequency that can be accurately recorded is half of the sampling rate.

Since the human ear can detect frequencies all the way up to 20 kHz, 44.1kHz allows for frequencies up to 22.05 kHz to be accurately recorded.

If you’re a fan of math and physics, you can check out the Nyquist-Shannon sampling theorem for the intimate details on sampling rates.

Of course, there are those who feel that higher sampling rates can be detected by the average listener, but there are also several studies to the contrary.

In my ever-so-humble opinion, unless you have world-class monitors and an acoustically perfect space (and let’s be realistic, most clubs do not qualify in this area) then 44.1 is a fine sampling rate for all but the most detailed solo acoustic recordings.

Sample Resolution

Sample resolution, sometimes referred to as bit depth, is a slightly different story.

The number of bits that are used to store the value for each sampled voltage determines the resolution.

That is, the more bits you have, the greater the range of numbers that can be used to describe a sound, since each bit increases the range of values exponentially.

For example, one bit has two values, two bits have four possible values, three bits have eight possible values and so on.

Here’s another analogy: If you’ve ever used a CD deck for DJing, you’ve probably noticed that the BPM counters aren’t always correct.

One of the reasons for this inaccuracy comes from the fact that the counters do not include decimal places, so if you have one track that’s 125.1 BPM and another that’s 125.4 BPM, the CD player’s detector will round the tempo of both tracks to 125 BPM.

Trainwreck time.

Sampling resolution behaves in a similar manner.

When the original audio fluctuates slightly between two values, a sampler has to round these values to the nearest available number, so the more numbers you have available, the more accurate that rounding function will be.

What this means in plain English is that higher bit depths have better dynamic range, which means that the distance between the loudest sounds and the quietest sounds can be even greater, thus the quality of a 24-bit recording is perceived as being “better” than a 16-bit recording.

Of course, all of this goes right out the window as soon as the track is mastered for a club, since compressors, limiters, maximizers and such actually reduce the dynamic range of a recording so it sounds “louder” overall.

The video below does an excellent job of explaining how the mastering process has “evolved” over the past twenty years:

Again, a noisy venue and a bunch of illicit substances will limit your perception of these distinctions, but you should still understand all of this stuff anyway, since it applies to the techniques used to edit and process samples.

Those are the techniques that we’ll discuss in the next lesson.

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