This may be a bit offtopic, so I'm sorry if MG isn't the place for this questions. It turns out I'm not only new to modular but to music production too, and there're some basics I still don't get. I hope someone can point me in the right direction.

The issue I'm facing is about sound quality in general. When I record my music it doesn't sound as clean or proffesional as I would like. My setup is quite simple at the moment, this is what I'm doing:

  • One MI Plaits sequenced by a Minibrute 2, the out of the Plaits connected to the master input of the Minibrute so it gets mixed with the main out.
  • The main output of the Minibrute goes into a desktop mixer.
  • I also have one Drumbrute connected to that mixer.
  • The main output of the mixer is then connected to the computer line-in for recording.
  • In the computer I'm using QuickTime player for audio recording and I tried adding some compression with the Audicity software too.

I think this are some of the problems, but I'm not completely sure and I don't know how to fix them:

  • I'm using a cheap mixer and it may be adding some background noise. Well, this is easy to fix, get a better mixer, but... is it my main problem?
  • Using the computer line-in may be adding even more background noise. Do I need some kind of controller o external gear to properly record my music?
  • QuickTime player doesn't sound to be the best option for audio recording, should I use something more like Ableton? is there anything simpler? I've never used Ableton and it looks pretty complex.
  • Audio compression and equalization: I really don't get these two, I tried adding compression because it seems like everyone is doing it, but what is it? how does it work and how does it affect sound? Regarding the equalization: how can I get it right? how can I know if the mix is well equalized? how can I mke sure it sounds right in different headphones or speakers? and What gear or software should I use for equalization and compression?

There may be more problems with my recording process and I'd appreciate if you guys could point me in the right direction. Also feel free to suggest articles, videos, books or anything else that you think could help me understand the basics of sound recording.

Thanks!


OK, let's answer some of this, sort of out of order...

Not only will a cheap mixer cause noise, there are definite sonic differences between something cheap and something that costs more. And the difference there comes from component quality. Cheap stuff (like Ammoon, Alto, Harbinger, et al) cuts corners on components, with the result being looser tolerances, which sort of cascades as your signal path goes through the board. One sub-par component is bad enough...now consider what a couple dozen of them in an audio chain will cumulatively do. Plus, certain mixers have a very specific sound quality to them, most notably the English-designed/made ones. This is what makes a pre-Behringer Midas desk so desirable...but not so much a post-Behringer one, as these don't have the same rounded "English tone" anymore.

Computer line-ins aren't the right thing to use, nope. The culprit here is noise at your A-D conversion stage. This is due to the A-D on a typical sound card (or sound card on the motherboard, depending) being typically unshielded from in-case electronic noise, plus the fact that that connection is going to be a consumer-level (-10 dB instead of +4) line-in and it's also unbalanced, which tends to allow more electronic crud into your signal chain. The line-out isn't as problematic, but to get really good results on recording, you need an outboard interface that's +4 dB, takes either XLR or 1/4" TRS balanced lines, and has a proper ground. And one other point: everything in a recording setup should be star-grounded. By this, I mean that everything you use needs to have a ground that is the same as all other devices, usually done by grounding everything to a single ground point (hence the name). By doing this, you can lower noise and help avoid ground loop issues.

QuickTime is not only the wrong tool, it's also VERY out of date. Use a proper DAW. You already have Audacity, so try recording in that instead. I actually multitrack in Ableton 10.0.6...but I chop loops and clips and also do my final editing and normalizing in Audacity. It works better for that, while Ableton works beautifully on multitracking, track comping, and so on. Ableton is also not the only choice; you might look at Bitwig, which is similar but has some of its aspects more streamlined than Ableton Live.

Now for the last pile of questions...first, EQ. Technically, there's three types: parametric, graphic, and program. Parametric is the type where you can specify the frequency per band, the level at that frequency, etc; you often see these on mixing desks in some form or another. Graphic EQs are the ones with fixed frequency bands with level controls, and tend to see more use in live applications for room correction, but can also be useful for similar purposes in the studio. And program EQs are things such as Pultecs, where you have specific boost/cut stages with their own tailored frequencies, often also working on the overtone spectrum of the selected frequency. This last bit is very typical of the Pultec EQP-1A's low end cut/boost control, where the 'boost' also works on the overtones of the selected frequency, but the 'cut' acts like a normal shelf, with the -3 dB point at the frequency.

For the most part, a program EQ is the only EQ you should boost levels on. All other equalizers should be used to subtract from what's present in the raw signal unless you're using the EQ as an effect in some way. The reason for this is that it's easier to compensate for lower levels of something in a mix than it is to correct levels of some type that're too hot. For example, let's say that one track has a band in the lower-mids that's sticking out, a sonic 'lump' as it were. It would be easier to isolate the 'lump's' frequency and reduce that on that one track than to bring everything else up in various levels and bands to even out the 'lump'. But with a program EQ, what's being done is more akin to "sculpting your mix's tone color"; accordingly, most of the time you'll see program EQs on the final mixbus to do those timbral adjustments.

However, tinkering with EQ without a good monitoring chain...flat, unforgiving response from as low as is feasible in the bass all the way up to the ultrasonic...is basically pointless. It's like trying to read a map, but you've forgotten to put on the reading glasses you need...ergo, you're probably going to get lost. Never skimp on monitors...unless, of course, you're trying to check your mix on a more "real-world" equivalent, in which case you need to incorporate those "everyday" monitors alongside the other, more precise ones. And this, btw, is how you check your mix; if you need to know how something sounds on, say, a typical set of computer speakers, by all means use some of those after you've done your mix on the mains. But if something needs fixing as a result, do that work back on the mains again. Motown studios always had a pair of 6"x9" car speakers in some cobbled-together wooden boxes in their studios specifically because Berry Gordy wanted to know how their stuff sounded in your typical car...and of course, Motown stuff sounds great in the car because of this "check". Headphones, however, are not something you mix in unless you're specifically mixing for headphones.

As for compression, there are again several types. Limiters basically "smash" everything above their threshold level and hold the dynamic limit right there. More typical compressors have various (and often adjustable) settings for how aggressively the compression happens as the desired level is approached, plus what sort of degree of compression (ie: ratio) is needed. And program compressors, like their EQ counterparts, are more for riding gain and "gluing together" a mix while used on the mixbus. As for the right way to use these, first keep in mind that anything over 4:1 ratio winds up behaving and sounding like limiting, especially with a hard "knee" (that "aggression" setting) at the threshold level. To get a compressor to behave transparently, use lower compression ratios and softer "knee" settings, which will then allow the compressor to compress over-level signals enough to fix level problems but not to make the track in question sound like its being "mashed". Unless, of course, that's what you want, since compressors are also useful for adding distortion and overload character to sounds that can use a beef-up.

Program compressors, though...those are a bit different. In their case, you use compression to get the overall stereo level on the mixbus to "float" around the desired track's loudness without exceeding 0 dB. So the meters on a program compressor might be floating at around -3 to -5 dB, and you'll use the makeup gain to bring that result's level up to where you need it to be post-compression. These are a good bit trickier to use well; like anything else in music worth doing, they require practice.

As for what to use...that's up to you, and what sort of sound you're going for. A good place to start, though, would be KVR Audio (https://www.kvraudio.com/), which has a trove of free plugins. You should, over time, be able to find the ones that work for your music and workflow...but again, this takes time, because in this process you're actually tailoring your DAW to be your bespoke recording "instrument".

Hopefully some of that is of use...


Thanks a lot Lugia, it is very useful and I'm sure I'll be re-reading it a few more times while I do my research on the topic.


There are entire professions devoted to recording and mixing (I'm in one of them). I don't think a couple of paragraphs are going to do it in order to get you where you need to be. I'd recommend taking the time to learn some theory and practical knowledge. The most important part is experience... and you only get that from trying and making mistakes. I made several with EQ and compression when I first tried home recording back in 1991.

My first recommendation would be to buy a decent audio interface. I'd shoot for something in the $300 range for a beginner. This will dramatically improve your results once you've mastered gain-staging. I would also invest in a DAW. Reaper is surprisingly affordable at around $50US for non-professional use. The great thing about a DAW is that you can EQ, compress, etc. inside of the DAW... save all your settings... and then revise your mix later on. You'll also find it valuable for learning and practicing.

A quality pair of headphones and if you can afford them a quality pair of monitors will help. Bear in mind that when using speakers, the room that the speakers are in will ultimately impact what you're hearing with your ears.


Hey thanks Ronin1973,

I've been looking around and it looks like something like the Focusrite Scarlett 18i8 would be a good place to start, what do you think?

And if I understood correctly I could plug my gear into the Audio Interface and then to the computer via USB, I wouldn't need the mixer at all. This way I could even record multiple tracks at once using something like Reaper, which looks great by the way, thanks for the suggestion.

Then, having the sound in different tracks would also be easier to equalize, right? let's say I record the mix using Audacity and all I have is one audio file, no tracks and so, could I still equalize it properly? I guess it's more difficult and the results wouldn't be as good.

Anyway, do the same concepts regarding equalization and compression apply when playing live without a DAW? do I need equalization modules and so on? or is it something completely different?


The Scarlett would be a good choice for recording. Here are some issues you may encounter: all audio interfaces introduce delay between the time the sound hits the interface until its recorded and played back through the DAW. This is called latency or sometimes referred to as lag. Most mid-range and higher USB interfaces offer the ability to mix the direct sound (your synth) and audio coming from the computer so you can monitor without lag. As long as you're not trying to listen to what you're recording through plug-ins IN your DAW this is an acceptable solution. You can get interfaces with very little delay (practically zero) but the cost goes up substantially.

The other issue is that Eurorack synth level versus line level (the level of operation in mixers and like gear). Synth level is hotter than line level. It's very likely that you can just adjust the input volume on each Scarlett input to compensate. Do your homework if this will work with the Scarlett. Ask around.

If you record each part on its own track (you'll need two inputs for tracks in stereo), you will have them isolated. You can process isolated tracks separately and is the best way to mix/edit your music. If you record everything as one stereo file you are basically stuck with what you have. Any effects (EQ, compression, reverb) is applied to everything in the mix... which is something that you probably don't want.

If you're playing live without a DAW, you'll have to decide if you want/need EQ, compression etc. You'll also need some way to provide them. I don't see too much of this inside of most peoples' racks. But they'll often use an external mixer that has EQ built in. There are several types of EQs and a reason to use each of them if the material requires it. Without getting into a big philosophical, technical diatribe, I wouldn't worry about it in the beginning. If it sounds okay, it is okay. Having EQ, compression will make your live mix better. But if you don't know how to use them properly (for now) it's more trouble than it's worth. The exception would effects like reverb, delay, chorus, distortion, etc. They are very important in defining your synth sounds and shouldn't be overlooked.

Next chapter... SYNC. aka how to lock your Eurorack tempo to your DAW's tempo for overdubbing more parts into the DAW for later mixing/editing. Do a little research and report back. :)

There's a lot to learn from your starting point and no easy way to explain it without you doing a lot of hands-on by yourself. You may... and I say may... want to subscribe to an online course regarding mixing and DAWs since there's so much to explain. You'll also get a better understanding about signal flow between your modules and your DAW.