Anatomy of a Soundboard

In order to understand how a DAW works, one should first understand in extensive detail what that software is mocking, a real analogue mixing console. 

Anatomy of sound boards 

While we use our StudioLive AR16 differently than a traditional sound board (we use it more of an interface than a mixing device), we should understand the mixer as a prerequisite to understanding the DAW. 

The purpose of a mixing console, whether it’s in the form of a DAW or traditional mixer, is simply to take the signal from multiple input devices (microphones, instruments, etc.), mix it together to portray all the sounds together.  Remember that the purpose of live sound “reinforcement” and recording is to just help all the instruments come together in harmony, or to make someone’s vocal portrayed in a way that’s smooth and easy to listen to, and our job as sound engineers it to paint the final picture that bring the song or radio commercial or video come together. 

I try to separate audio boards into sections and sub sections-


  • Input connectors 
  • Gain (along with pad, phantom power, phase flip, etc.) 
  • EQ (along with low cut) 
  • *On some boards: compression 
  • Aux sends 
  • Master Volume (pan, mute, group assignments) 


  • Output connectors 
  • Aux sends control 
  • Master volume control 
  • Monitoring 

Input Channels 

Channel Inputs and Types:  All the input channels can be broken up into mono and stereo input channels.  Mono channels accept a single signal that gets duplicated into 2 signals, one for the left and right side of the main mix.  Stereo signals accepts 2 signals that are kept separate, then the corresponding signal goes into the corresponding left and right sides of the main mix. 

All channels will have various input connectors for you to use.  Mono channels will traditionally have a single XLR input that is connected to a microphone pre-amplifier, so you can use these for microphones (or any device outputting at “mic-level”).  To accompany the XLR input, you may have a ¼ inch input for line-level devices.  Some boards will also feature direct-injection boxes built into channels so you can plug guitars directly into the board, as does our AR16.  Stereo channels usually use 2 different input connectors for the stereo signal, one for the left and right side.  These may be ¼ input or RCA inputs.  Occasionally, you may see a stereo input with a 3.55mm input that allows for a stereo signal in a single connector (although do not be mistaken, there are separate wires in this that allow for 2 separate signals).  I’ll get into unbalanced and balanced connection later on. 

Input Controls Each input channels has gain controls (sometime called “TRIM” or “input volume”).  This controls the volume that comes into the mixer from the input device.  The whole point of these it so that no matter what the volume of the signal that is coming in, we can make all the input signals to the board the same volume.  No matter if it’s a guitar going through a direct-injection box, or if it’s a vocal microphone, or a mic on a drum, we can use the trim to bring all the signals to the same volume so we can work with them on a level playing field, and so that everything is in a good range or signal for the electronics of the board to work with.  There is a proper level-setting procedure to follow when you set this gain control.  If it’s not done properly, your signal could distort or you could have unnecessary hum from turning the signal up elsewhere to make up for a gain setting that is too quiet. 

Level Setting Procedure 

Every time you set the gain control on any device (mixer, audio interface, etc.) you should follow these guidelines. 

  1. Verify that the signal is being dealt with properly.  If it is a condenser microphone, it needs phantom power.  Guitars need to either be recorded with a microphone, go through a DI box and then into the MIC preamplifier, or be plugged into the built-in direct box on channels 1 and 2. 
  1. Set the trim properly, as this is the biggest component of this process.  Do not just depend on the signal LED that is right next to the TRIM control, instead press the “PFL” button that is next to the channel fader.  (if you encounter an “AFL” button, this stands for after fader listen, and it will put the signal after the volume has been changed by the fader).  This will show the input volume (the volume PRE the fader control) on the metering stack.  Play your signal- if this is a singer they need to sing at the volume that they will when they officially record, etc.  On average, the signal should be coming in around “0”, with loud spikes in signal residing in the yellow, but never close to the red.Doing this procedure allows not only a proper signal level to be worked with, but also everything will be mixed at the same level. 

Channel Parametric EQ Each channel features 3-band parametric EQs.  On the mono channels, the mid control is “sweepable,” meaning you can control what frequency (the light-blue dial) the volume control (the dark-blue dial) is changing.  However all the high and low controls, as well as the mid controls on the stereo channels, are fixed frequency controls, so you can only change the volume. 

EQ is a huge topic and can be used in many different ways to serve many different purposes.  For recording scenarios and for what we’re recording, it is probably best to leave the EQ un-touched and EQ everything later on in the DAW.  Not only can you always go back and change the EQ, but you have a much larger selection of EQs in the DAW workspace. 

EQ can be used to correct problems with the recorded signal (although it is better to try to record in such a way that you won’t have these problems in the first place, but sometimes you have no choice).  EQ is mainly used in the recording space to help blend all the recorded things together and to make the sound pleasurable to the listener.  Also, when dealing with recordings you must remember that different devices play the frequencies at different volumes, so we also need to use EQ so that our program material won’t sound different on different listening devices. 

Also a part of the EQ section is the “Low Cut” control that is oddly, for some reason, located next to the trim control.  Both low-cut (aka high-pass) and high-cut (aka low-pass) filters cut off frequencies at the extremes of the frequency spectrum.  Low-cut filters help remove low-end rumble or pops from a vocal or just remove signal that is not necessary in the mix.  (if you ever run into an “octave” control,  this basically controls how frequencies near the selected frequency get cut off as well.) 

Auxiliary Routing Controls All mixers and DAWs have “AUX sends.”  Auxiliary sends or “busses” supplement the main output bus and provide us with a way to create another mix for some other purpose.  These mixes may be used to create a monitor mix for an artist so they hear the rest of the band, or to create an FX send to send sound to an external FX unit. 

In DAWs and mixers, send and be “POST fader” or “PRE fader”, sometimes these settings are hard set and sometimes they are controllable.  A POST-fader send sends signal after the channel volume fader controls it.  So if a POST-fader send is sending signal, the channel’s volume in that send will changed depending on the volume of the main channel fader.  POST-fader sends are favorable for FX sends, listening rooms, recording, etc.  PRE-fader sends send signal regardless of the fader signal.  So if you change the volume of the main channel fader, that channel’s volume in the send is unaffected.  This is favorable for stage monitors or headphones, etc. 

PAN/BAL Controls Mono channels have PAN controls, because you are taking a single signal and controlling the ratio that each signal that makes up the stereo signal gets.  Stereo channels have BAL (short for balance) controls.  This is different because unlike mono signal that get split up evenly for each signal for stereo, stereo channels already have dedicated stereo signals for the stereo output.  Because of this the “BAL” control just controls the volume of each side. 

Channel MUTE The channel mute just stops signal from going to the main stereo bus.  Muting always effects POST-fader sends, but whether or not it mutes signal in PRE-fader sends depends on the mixer, it’s different for different ones. 

Our board has a digital return channel that allows signal from the computer to come back from the computer or Bluetooth.  Also there are analogue options like the RCA inputs. 

Output Section and Controls 

Our AR16 has fader controls for the monitors sends.  For the FX fader, this controls the amount of FX that are returning into the main mix bus (as it has a BAL control).  The faders for Mon 1 and Mon 2 control the main volume for that bus and control the output signal for those.  Since they are master faders they have AFL buttons for motoring. 

Control Room Volume Controls  

Our board is specifically designated for recording and editing applications so it also features control room volume controls. 

Solo Master.  When you press “AFL” or “PFL” it reverts the signal from the main bus and sends it to the solo bus.  This is the volume control for that solo bus.  Leave this at unity gain, as we do not want it effecting the signal. 

Control Room.  This controls the volume to the control room outputs. 

Phones.  This controls the volume for the headphones output. 

 Built-In Effects Processor

This effects processor is an external effects processor built into the board.  It is classified at external because it is not a processor built into a channel, rather we send signal over to it and then it applies 100% of its effect to it and returns that back into the main bus.  (Internal effects processors are on the individual channel and therefor do not apply 100% of their effect.  You may also hear someone talk about how much effect is applied by referring to how “wet” a signal is.).  For studio purposes, I would use effects in the DAW. 

Differences between Sound Boards and DAWs 

DAWs add a main feature to traditional sound boards in that they can record signal for their channels and allow us to edit that signal very easily.  Their signal is the input for the channel.  Where on a normal board we would have input connectors, trim, etc., in a DAW that signal is the input. 

Also DAWs are very dynamic, in that we don’t have set processing on each channel.  For each channel we can use as many as whatever type of EQ we want, with as many compressors as we want, in whatever order we want. 

Also because DAWs are linked to their recorded signal, we can determine automation as the song or recording plays (we can program controls to change as the song plays). 

DAWs are pretty much traditional mixers other than that.  They have sends also, but we can have a bunch of them and control if each are pre or post and also setup our own busses to route audio. 

Workflow between the AR16 and DAW 

If we want to take a bus out of the DAW so we can hear it, we must use converters, which our AR16 plays the role of.  The AR16 has 18 input converters and 4 output converters.  The 18 input converters correspond to the 18 inputs on the board.  The first 2 output converters correspond to the digital return channel, and the second 2 optionally route to the last stereo channel (activated by the 3/4 button). 

Your DAW will have settings for these.  Usually input converts are assigned specifically to a channel, while output are assigned to a bus and that bus’s signal comes out the converter assigned for that. 

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