We get a lot of emails from people asking us how best to record their guitars, and since I happen to run a little recording studio in Los Angeles I thought it might be a good idea to start a series on the recording process. So here’s the first part of my series, in which I’m going to introduce the concept of signal flow, which is basically how sound travels from an instrument into a computer (or a tape machine, back in the day) and then eventually back out through speakers. The better you understand how signal flows from the source (instrument or voice) back to your ears, the easier every step of the process will be along the way.
I’m the first to admit that I’ve probably written way too much here, but I didn’t feel right leaving any of this out. At the bottom of the article you’ll find a few points to get you started if you really don’t care about any of this and just want to get started.
A lot of people like to use the analogy of plumbing when discussing signal flow – if you think about water getting from point A to point B you realize that there always has to be a pipe for the water to flow through and a series of taps that determine when water flows, doesn’t flow, or comes out of a tap. Audio signal is basically the same – it’s an electrical current or digital signal that needs to flow from point A to point B through a series of taps, etc.. and your computer and software will determine whether those taps are open to the signal or not and how it gets routed from the microphone to the speakers.
There’s an awful lot going on between the time you create a sound for a recording and the time you hear it played back from a speaker, so if you want to learn how it all works you’ll want to bear with me. It sounds complicated at first, but like most things in life there comes a point when suddenly it makes all the sense in the world, and if you want to record music you’ll want to know this stuff
For our example we’re going to take a guitar played into a microphone going into a DAW (Digital Audio Workstation, which is audio-speak for an audio program like Pro Tools, Logic or Garage Band) and back out to a pair of speakers or headphones. This is how millions of recordings are done these days, and I’ll assume you’re not working with a mixing console, patch bay or outboard gear (since if you have those things chances are you know what they’re for and how to use them).
The basics are pretty straightforward, but the really important thing to get here is what happens on the way in (signal getting into the DAW, or what we might think of as the recording process), versus what happens on the way out (mixing, processing and playback). Along the way you’ll want to understand monitoring, which is what we call listening to yourself through headphones as you’re playing.
The first thing you have to do is get the sound into the DAW, which means that physical sound (which is analog) has to get converted into digital samples so that your DAW can understand it. This is the role played by your audio interface. If you have a Digidesign (now Avid) M-box, or an M-Audio something or other, or anything that you plug your mic into that then connects via USB or FireWire into the computer, this is your audio interface, and it is doing the conversion of analog to digital for you (we refer to this as a/d conversion, and when the digital has to be converted back into analog sound we call that d/a conversion, logically enough).
[I’ve intentionally skipped a step here, because many interfaces have built-in mic pre’s (microphone preamplifiers) – a microphone’s level is relatively weak and needs to be bumped up to a level that a mixer or a DAW can deal with, and that is the role of the mic pre, and mic pre’s can really affect the sound of your signal (after the microphone itself they may be the most important part of your sound) but since so many audio interfaces have built-in pre’s I’m not going to spend a lot of time on that here.]
If you have a USB microphone the conversion happens in the microphone itself and sends a digital signal directly to your computer via the USB cable, so you may be able to operate without any interface at all.
Once the signal is getting into your computer, your DAW needs to know where to look for it. The sound coming into the DAW is called the input, and somewhere in the preferences of your DAW you’re going to tell it where to look for incoming signal. If your audio interface can handle more than one input at a time, you will then have to tell your track which channel to look at for input. Since most smaller interfaces only have two inputs, chances are you will simply have to choose between inputs 1 and 2, but on bigger systems you will have to be on top of exactly to which input you’re sending your signal.
Once you have all of this taken care of you can hit record and track audio in your DAW (tracking is just another word for recording). If you are listening over headphones you may notice that what you hear comes out of the headphones slightly later than it goes into the microphone, and this is called latency. Latency is basically the time difference between the sound going in and the sound coming out, which makes sense if you stop to consider that between input and output the sound has to travel into a microphone and through a cable or bunch of cables, at some point get converted into digital samples, be read by a computer program and maybe get processed a little before output, get converted back into an analog signal and then through a cable into your headphones. One of the big differences between the bigger pro rigs and the smaller home setups is how they deal with latency.
In this type of setup you will be recording a clean, unprocessed signal, and any effects you want will be added to the signal in the DAW, and it is important to understand that they will be added on the way out. This is important to understand, especially when it comes to plug-ins. Plug-ins are virtual effects (well, they’re real enough, but they aren’t physical pieces of equipment, so we call them virtual). They can add reverb, EQ, delays, or any other effect you might want. They’re easy enough to use if you want to process audio you’ve already recorded, but they can be extremely tricky if you want to hear yourself through the headphones with the effect while you play. For now, I recommend you not concern yourself with plug-ins until after you’ve recorded, or until after you understand how signal flow works well enough to play around with it.
Once you’ve recorded your audio and you want to hear it back or mix it, you’ll want to understand a little about how your DAW works, and since pretty much all DAWs are just virtual mixing consoles, you’ll want to understand a little about how a console works.
One way to think of console is as a summing machine – it takes two or more tracks of recorded audio and adds them together to give you a single stereo track (of course you can ‘mix’ a single track by processing it and altering the audio in all sorts of ways and then printing or bouncing that track). But before you send the tracks to the mix bus (also called the master fader or the 2-mix) you can play with levels, with panning, with EQ and effects and with a few other things, and that all falls under the heading of mixing, which we won’t get into here.
For signal flow, the important thing to understand is that within a console (or DAW) each track represents what we might think of as one ‘voice’, and that music goes from the track through any processing we might do (via inserts, because you insert an effect on the track) to the master or mix bus, then back to your audio interface’s output. Again, since most home audio interfaces only have two outputs (right and left or 1 and 2) we won’t get into that.
The last important thing, of course, is to hear the audio, For this you’ll want to make sure that your program (or board or mixer) is sending the output to the right place, which will either be your computer’s sound card or your audio interface. If it’s your computer’s internal soundcard you probably already know how to hear the audio, and if it’s your interface then you’ll have some headphone outs there or some main outs to send to speakers.
***Some things to look for if you’re completely new to this, have plugged in your mic, and are frustrated that things aren’t working***
I/O stands for Input/Output, and you’ll want to check your computer’s and/or interface’s I/O settings to make sure that the audio is going into your program and coming out in a way that you can hear it.
1 – Make sure that your computer and your DAW recognize the USB microphone or the audio interface. This will likely be in your computer’s system settings and your DAW’s preferences or audio/hardware settings.
2 – You can’t record without a track to record onto, so the first thing you’ll want to do after opening your DAW is to create a track. Chances are your DAW will have a ‘track’ menu and you’ll look for a menu item called something like ‘add new track’ or ‘create track’.
3- Each track generally has a little meter so you can see your levels. If you don’t see anything happening on the meter, chances are your I/O settings aren’t right.
4 – I/O settings: For some stupid reason, this is often not as easy to find as it should be, given how important it is. You’ll want to look first in preferences to see that the DAW recognizes your mic and/or audio interface, and then for menu items that might be called ‘I/O’ but might also be called things like ‘hardware’ ‘setup’ etc… If you have the manual for your DAW look for I/O, Input, Output, setting, etc…
5 – Once you have audio coming in to the DAW (you can see that the track is receiving signal on the meter) you should be fine if you can hear yourself, but if you can’t, then it’s back to #4 to make sure your Output settings are correctly set either to your computer’s sound card or your audio interface.
6 – You’ll find videos on YouTube explaining the basic setup and operating procedures for any DAW you might be working with, so if you get frustrated that might be a good place to look for info that’s more specific to your program.
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