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Working with Internal Audio Routing - Part 1

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This post is part of a series called Working with Internal Audio Routing.
Working With Internal Audio Routing - Part 2

Trying to move audio data from one source to another should be a very straightfoward process. Simply setup your I/O to send one sounds output to the input of another. In fact, we do this all the time inside our DAWs from track to track. But what happens when we want to route audio between applications?

Say you want to show a mix you are working on, in real time, to a friend over Skype or Teamspeak. How can you do it? The truth is, you can easily route audio internally between applications on either PC or Mac, as long as you have the right middle-man software.

Interface Mixers and How it all Works

Many people forget that their audio interface more than likely has an audio mixer application that can be installed alongside the drivers. Even if it is installed, why bother touching it? Typically they do not let you record, mixing is limited, and adding effects can be sketchy.

Sure, you can do live almost-zero latency "mixes", but if you have to open your computer to do it, why bother with these specialized interface mixers?  Because most of these interface mixers let you have direct access to the internal routing of the interface.

While there are various programs to assist with internal routing (we will go over particular cases next time), if you are running a pro audio setup, chances are your audio interface may already support the necessary routing. The easiest way to see if your interface supports internal routing (besides reading the manual) is to check for a Mix L-R Return (or some other logical naming scheme) option on your audio inputs.

When looking for these inputs, they typically will appear after your analog and digital inputs. For example:

  • Microphone 1-2
  • Microphone 3-4
  • Mix L-R Return

In most normal applications, these inputs will allow you to route the primary outputs of your interface back into your DAW before it ever gets converted into analog. Think of it as an Aux Send for your interface's main output.

While on the surface it may not seem that useful, this is the feature that lets you route your main outputs from one program into another. So when would you use it? A common usage for this feature would be if you wanted to have mixing session with a producer or artist who could not be physically present, but could be on Skype or some other VoIP service.

The alternative is to go back to the days of analog telephone patches. Your choice. I personally cringe at the thought.


For those of you who want to try it, the process goes like this:

  • Open up your DAW of choice and particular session.
  • Open up your VoIP program, such as Skype.
  • Create a new track in the DAW, and route your microphone in for talkback.
  • Go into your interfaces dedicated software mixer (Universal Control, CueMix, MixControl), and insure that computers playback is routed into this mixer (usually via a drop down menu or track input on the interface mixer). For example, CueMix (MOTU interfaces) has a drop down option File>Mix 1 Return Includes Computer Output, and MixControl (Focusrite interfaces) has a DAW option under each track input.
  • Inside your VoIP program, change your Microphone or Audio Input settings to the Mix L-R Return.

As you can see, the overall process is not very complicated. It's simply a matter knowing the capabilities of your interface and built in interface mixer. If you are fortunate, enough have more than one Mix return (maybe a Mix 1-2 Return and a Mix 3-4 Return), then you have even more routing options available.

But with all good things there's a possibility that big problems will arise.

Avoiding Problems

While the above seems very simple (and it is!), there are a few pitfalls you need to be aware of. When routing audio back and forth, there's a chance you'll cause some nasty feedback loops, or at the very least, really annoying audio doublings that are a few milliseconds off, and will cause a fluttering effect.

These issues arise because we are taking an auxiliary feed of our main outputs, which (thanks to the ability to include all the computers output) take in all of our combined audio signals between all open audio applications.

In order to prevent feedback, we first need to know exactly where it could come from. In this situation, the only audio signals being fed into our computer are our own vocal microphone and our partners VoIP feed. Obviously our mic signal won't creating the feedback unless we smack the microphone right up against our speakers. So the problems will come from the VoIP feed.

Here's what can happen:

  • our partner speaks into their microphone,
  • the signal is transferred to our computer via the VoIP,
  • our interface mixer takes into account all audio going to the Main L-R (which includes our partners voice),
  • the voice is fed back to our partner,
  • and they hear an annoying duplicate of themselves.
  • Worse yet, if they are doing their own internal routing, they might begin to get feedback in their system!

The easiest way to correct this problem is to use separate audio outputs for the incoming VoIP signals. For example, if your interface has multiple outputs whether for multiple speakers or headphones, simply specify the incoming VoIP to go to those outputs instead of the Main L-R. If you are using a headphone jack for your monitoring, you probably won't need to do anything else (assuming you are sending your Main L-R to the headphones).

If you don't normally use your additional outputs, you may need a separate physical connection back to your speakers via a line level mixer or multiple input jacks. While it is a little more convoluted, there are no real alternatives if the problem arises!


Conclusion For Now

As you can see, internal routing to VoIP, while fairly straightforward, can get a little hairy. If something is not working, take a step back, start from square one, and follow your signal flow. Draw a flow chart if you have to!

The other issue you need to keep in mind is VoIP compression. Since bandwidth is not unlimited, the VoIP will have to do some compression to deliver your mix to the other side. While it is surprisingly-decent sounding, VoIP is not something you will want to do critical editing/mixing on, but for general levels and overall feel of a mix/track it can work just fine.

Next time we'll look at various programs that allow us even greater audio routing flexibility inside our computers.

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