It’s bad to start your post with an apology, but I must: this one won’t deal with FreeBSD as much as set some foundation for the posts to come.

The base for a good digital audio workstation, or DAW for short is audio interface. That is the piece of equipment that will do the most demanding task of converting from analog to digital and vice versa. As audio we hear is, after all, analog one must pick it’s audio interface a bit more carefully than the rest of the gear, so here are my tips on choosing a decent one:

  • find as much as you can about it’s ADC (analog to digital converter) and DAC (digital to analog converter)
  • match impedance (more on that later)
  • as high sample/bit rate as possible with internal mixer using more bits than ADC/DAC so it has room to handle clipping (common these days is 24/32 bits)
  • ability to be world clock master and slave (more on that later)

Impedance is a fancy word for electrical resistance. It has to do with the fact that resistance of a device is not the same in all circumstances. One of those situations where this really matters is equalizer: one band actually has lower resistance for a certain frequencies, and higher for others. All I’m trying to emphasize here is that when you here “impedance” you should think “resistance” and keep in mind that it’s dynamic.

One thing all electrical circuits like is when impedance of it’s output matches the one on the input of the next step. When I say “like”, I mean least amount of energy is wasted in transit from one circuit to another (read: you get more signal/noise ratio) and the least amount of distortion is introduced (unfortunately, every device adds some distortion). So, to have a perfect audio interface, choose the one that has mic, line and hi-z inputs. Mic input should have 48V option which is needed for condenser microphones (studio microphones). Line is what most devices use, like mp3 players, other sound cards and synths. Hi-z is just a fancy name for “guitar input”. What you should look for with hi-z is a active/passive switch. Active pickups have small amp inside them and need battery, so they are easy to recognize. Passive pickups are the ones without battery, and they have 3 to 9 times lower output than active ones (depending on the chosen pair of active/passive pickups).

As digital audio IO must operate at the precise same frequency across all devices, once you get guitar or vocal processor, you’ll need to sync your audio interface and processor. There are multiple ways for achieving that and it mostly depends on the way you’re going to connect the devices, but let me explain why it’s important. All digital devices use “the clock”. It’s what tells them “hey, it’s time for the next sample” among other things. That clock is usually quartz crystal which has a property of oscillating when electric current is introduced. When you have two devices with their own clocks, they have slight differences in frequencies which come from slight differences in crystals inside them. You might think “I don’t care about few milliseconds” of delay, but that’s not what’s in stake here. If digital device misses the clock beat, all audio can become gibberish and noise. This is solved by having devices that can use external clock as it’s own. Obviously, one of the devices must “export” it’s internal clock (acts as master) to other devices (slaves). S/PDIF and AES/EBU digital connections can also transmit the clock but you have to check your devices for such capabilities as not all can work this way. The safest option is to have World Clock on all of your devices, where your audio interface is the master. World Clock connector is BNC. It is especially important when you want to connect multiple devices to your computer (for example, via USB) as otherwise you’ll get a lot of small errors known as jitters.

For someone starting with audio, this must sound terribly boring, and maybe too technical, so please leave a comment if I didn’t cover something properly, or you have an idea how to make it more fun.