If you work in professional audio, today, you are likely using a whole lot of digital equipment. Most new mixers have some level of digital processing built in. PA control and manipulation is typically digital. Even wireless microphones are transmitting digital audio signals rather than analog lately. Every time an audio signal enters a piece of digital gear there is an analog to digital conversion which occurs. Every time a piece of digital gear puts out an analog signal the conversion happens in reverse. In this article we will concern ourself mainly with converting analog to digital.

Getting up to speed

Audio, as we encounter it, has two major forms. Analog, and digital. The definitions of analog and digital, according to google, are as follows:

Analog

ˈanlˌôɡ – Adjective – relating to or using signals or information represented by a continuously variable physical quantity such as spatial position or voltage.

Digital

ˈdijidl – Adjective – (of signals or data) expressed as series of the digits 0 and 1, typically represented by values of a physical quantity such as voltage or magnetic polarization.

Analog Audio – Sound is vibration. Vibrations in the air result in the propagation of tiny pressure waves through the air. The tone of the sound is determined by the frequency of those repeating pressure fluctuations. These waves travel through air much the same as wave propagation in the ocean. The intensity of the vibration determines the amplitude of the wave, technically known as the sound pressure level (SPL). Which is measured in Decibels (dB). These waves need to be converted to electrical signals before they can be translated to a digital format.

When the aforementioned pressure waves make contact with transducer in a microphone, that acoustic signal is converted to analog audio. The membrane of the microphone vibrates at the same rate as the acoustic sound, moving a magnet through a coil, and creating minute electrical fluctuations. These electrical fluctuations result in an electrical representation of the sound, and allows that signal to be sent over distances through copper wires for processing and amplification.

Digital Audio – Once in analog form, the electrical signals can be converted to digital ones and zeros. This is accomplished through sampling (or pulse code modulation). In sampling, the computer takes frequency and amplitude information from the electrical signal at constant intervals. You could think of this as a frame rate in film. One piece of this snippet is useless, but all of them in succession allows for the digital storage and recreation of analog sound information. Of course some in-between information is lost, but if sampling is done in abundance, that loss of information is negligible.

The conversion itself, how does it work?

An analog to digital converter (ADC) takes samples of an analog audio signal over time at a specified rate. This is known as the sample rate. Each sample is an amplitude snapshot of that moment in time within the sampled range. Because samples are momentary snapshots of time, there is inevitably information that is lost between samples. Converters use a process called quantization to give numerical values to each point in time, and also rounds to fill in the gaps between samples in order to replicate a constant signal. This inevitably produces quantization errors, which can result in a higher noise floor. It is widely believed that this space between samples results in a stair-step recreation of the original sound which cannot possibly be an exact recreation of the source signal. The only undesirable is the noise floor from quantization. This can be proved with a combination of oscilloscopes, tone generators and spectrum analyzers (See video below).

The bit depth is how many bits are in each sample. With more bits to work with (i.e. 16 bit or 24 bit) quantization is smoother which results in a lower noise floor, ultimately giving the recording a greater dynamic range. An 8 bit conversion can produce artifacts which are perceived as a buzz. These are the true result of quantization errors. This can be alleviated by random noises called Dither. Dither will not give you back your dynamic range, but, when added pre-conversion it will smooth out that harmonic distortion and give you a flat noise floor, rather than buzz. When gained up it sounds like tape noise or hiss to most, but is much more aesthetically pleasing than an abrasive hum.

Nuances of Digital Audio Conversion

The Nyquist Theorem

There is a limit to the frequency range which can be converted in an ADC

The highest convertible frequency is always half of the sampling rate.

Frequencies above the Nyquist frequency can still be interpreted by a ADC

ADCs will reinterpret those inaudible signals within the convertible range.

This results in undesirable harmonics and distortion which weren’t in the original analog signal.

This phenomena of reinterpreting higher frequencies is called aliasing. All ADCs today have anti-aliasing filters, which allow only the desired frequencies to enter the converter. So an ADC running at 44.1k is converting frequencies above the 20k hearing limit of human beings. But the frequencies between 20k and 22.05k are there for the use of the anti aliasing filter and is not to be converted with the rest of the information (between 20hz to 20khz).

Why do I care?

My Personal Take

Find yourself a high quality and a low quality converter. Take any audio source, I recommend an SM58 and a simple, transparent preamp. If you split the signal after the preamp and record the same source with both of those preamps, often times the lower quality converter will have either a very thin, or an undesirably muddy in comparison to the higher end which (should you have true quality on your hands) will have clear and transparent representation of all frequencies. Anti aliasing filters are delicate things. It seems as though a low pass filter is all that is needed, but simply truncating your range close to frequencies that you do want can have an adverse effect on the overall tonality. A dull conversion might have a low pass anti aliasing filter with a slope that starts cutting as low as 15k. A filter set too high can distort lower frequencies, as information above the nyquist frequency gets reinterpreted to the low frequencies.

If you really want an in depth look at analog to digital conversion, view the video below. It is easily my primary resource for this article. What I’ve written above is just ment to scratch the surface, if you’ve read this and find a fallacy think I have left out something important. Please comment or message me. This blog exists for learning!