Getting the best sound you can when playing live could require a little more work than a tweak here and there on your EQ knobs. If you use music from a variety of time periods, genres, and sources, you may experience noticeable disparities in tonality, loudness, and general quality. Luckily, there’s something you can do.
Final stage compression and EQ can totally transform the sound you get from your system, and automatically level out different sounding tracks – and the best news is, you can dip into the basics right now.
THE MISSING LINK
You may be lucky; if you use Ableton Live, Torq, or another piece of software that can host VST/AU plugins, this step isn’t necessary. If you’re using Traktor or another piece of kit that doesn’t, though, then you’ll need to read the rest of this section.
When we get to the input stage in a minute, all you’ll need to do is select the Jack output as the input and voila – a Jack shaped tunnel through your system.
You’re now ready for the next step.
MASTERFUL POTENTIAL
Time to look at what we can actually do with our sound! Here are some basic terms you’ll need to know:
- Compression is a process in which peaks in loudness are reduced to create a more even volume in an audio signal. Compressors work by reducing any audio that breaks the defined threshold by a ratio of itself. The master gain is then increased, and because the mean volume can now be higher, as the peaks are no longer as pronounced, the signal can be louder and more uniform.
- Limiting is the logical conclusion to compression, whereby the ratio that the peaks are reduced by is infinite and so essentially the audio is squashed below a maximum loudness. This ensures that no audio can ever reach above a certain level and so is great for matching volume throughout a set, especially if the limiter has an ‘auto gain’ function that ensures the dB level is always pushing against that maximum line.
- Graphic EQ is probably familiar to anyone who bought a stereo in the 90s (or has ventured into their iPod’s EQ setting, at least). A number of frequency bands at predefined points are available for tweaking and can be used to stamp character onto audio and adjust for how a room affects sound.
- Parametric EQ is a form of EQ where the bands can be freely set to any frequency, and their bandwidth can be played with to be as wide or narrow as you choose. These are really good for sorting out a room’s acoustics.
- Amp/Tape/VCA/etc Emulation is a huge buzzword in the digital audio world, because for all the advantages that digital audio brings the thing that’s missed the most is the character that analogue equipment can bring to audio. The above processes can be done absolutely perfectly in digital, but with analogue processing there’s always some kind of colouration. Especially when compressing a signal, saturation occurs. This is when pleasant harmonics are created as the signal overloads and folds back into itself, whereas digital overloading is just a mess.
It takes practice to get good results from mastering tools, but the key is to be subtle. The more extreme your compressor threshold, the more the signal will be affected even as overall volume takes a dip – and the greater the ratio the more squashed things will sound.
Which way round you route your EQ and compressor is up to you and your experimentation. If you EQ first, it will affect how the compressor works and with practice you can use this to your advantage – pushing up the bass in a signal will have the compressor kick in on that bass and can give that ‘pumping’ sound. EQ after compression allows you to be a little more precise and ‘even’ sounding, but you will still have to be mindful of peaks getting too pronounced.
Of course, there are a huge number of ways you can manipulate your sound – as you’ll no doubt discover as you start to dip your toe in the world of audio plugins – but I’d recommend you keep things simple and stick to compression and EQ for this task, with a little warming if that’s the sound you’re after.
STAND ALONE, OR UNDER MY WING?
When it comes to the software required for the actual mastering process, there are two main options: a plugin host or standalone software. If you’ve never run across plugins before, they are a pretty simple concept to grasp; software is often created with a mind to expansion by third party developers, who can harness a bridge that allows them to tap into the main software and augment its capabilities with their own software that – you guessed it – plugs in to the host. Computer music has a long history of plugins, and the two biggest types of plugins are VST (Virtual Studio Technology) and AU (Audio Unit), popular on PC and Mac respectively. Because a plugin can’t run standalone, you will need a host to use one; because a full DAW like Ableton Live, Cubase, or Logic is overkill for our purpose, a specialised plugin host that just hosts plugins is more suited – and as an added bonus we found a free one for Windows, VST Host, and a reasonably priced one with a free demo for Mac, Rax. If you do decide to go for the more fully fledged system, Reaper is always available at a very reasonable price.
There are so many plugins we couldn’t possibly make a definitive list, but a couple I like are PSP Vintage Warmer, which sounds excellent and really warms up and smooths out audio, as well as the VC76 from Native Instruments’ Vintage Compressors collection. EQ wise, the Fabfilter Pro-Q has absolutely excellent zero latency sound.
Most paid for software have demos you can try, but we scoped out some free options for you too – give Blue Cat Triple EQ, a ‘semi parametric’ EQ tool, Voxengo’s GEQ, a graphic EQ that adds colouration to your sound, and Molot, a low latency compressor a look (although your Russian might need brushing up, it does sound good). If you’re prepared to do a little searching of your own, KVR Audio has one of the biggest databases of standalone software and plugins on the net – if you really want to sound individual, it’s worth putting in the ground work (of course, if you’re the caring sharing type, why not let us know your finds in the comments?).
Standalone software is out there, and we’d suggest you take a look at IK Multimedia’s T-Racks and NI Guitar Rig. Each allow you to mix and match various different models of effects processors within their systems (like the aforementioned Vintage Compressors for Guitar Rig), and they both have low latency modes for direct playing as well as add ons and light versions.
LIMITATIONS
Perhaps the biggest limitation here is that all this relies on you mixing in the box. If you do your mixing in an external mixer then you won’t be able to apply these stages to your final output without running a cable back into your audio interface and then outputting from your audio interface to speakers… and that’s probably going to stretch the acceptable boundaries of latency. You’ll still be able to do it per channel in order to level things out before they get to the mixer though. If your software is like Itch or certain Traktor LE versions that lock the software to the audio interface, unfortunately you’re out of luck for this guide and will have to look into dedicated hardware if you want to give this a try.
Another limitation is with the software itself. A lot of DJ software is closed box because a large amount of R&D goes into creating the lowest possible latency and CPU saving processes to ensure a great user experience, and you might find that the better a plugin sounds, the more latency it introduces into the signal and the more CPU it eats up. This is just an unfortunate reality, and it’ll take some trial and error to find the perfect setup.
Finally, by the very nature of what you’re doing you may lose a little control over the mixer’s EQ, volume, and effects as the compressor works against them and keeps trying to make things smooth. Worse still if you really set things up wrong the sound will be completely flat and lifeless. For this reason it’s best to employ this technique subtly: Practice makes perfect.
Check out this video of Mr Scruff – pay attention to the latter part where he shows off his hardware final mix EQ and gain stages and gives his reasons… impressive!
examples here https://soundcloud.com/wolfangelocruz
Not recommendable, compress and limit music even more? when it’s already lacking dynamics.
Protip: Good quality music (and profesionally mastered) + a real decent sound system, like Funktion One = transparent good sound.
It’s better to have nothing to correct in the first place.
just to add:
SAVIHost is a little derivate of VSTHost that
has been crafted for the sole purpose of automatically loading exactly one VSTi.
Yes but I think he is suggesting older tracks that are not squashed to oblivion will benefit. A lot off pll used to stick Vintage warmer at the end of the master in live when mixing in that.
Would you recomend using a apogee duet to improve the sound ?
Super Duper Post! Thanks!
Timecode + Jack Audio = no go
These videos are spot on – Highly recommend checking them out.
These videos are spot on – Highly recommend checking them out.
Sorry, but this is terrible advice – dance music is already over-compressed as it is, and if a track has been mastered properly it shouldn’t need any further compression or EQ. Any final EQing or limiting on the entire mix should be handled by the engineer responsible for the club’s sound system, not the DJ.
they’re talking about a mix to be recorded and posted online, not for live use.
these recommendations really are overkill. If you are playing a place where consideration of the room really is an issue then there will be in an in house sound engineer who will manage EQ and compression for you as the last part of the signal chain. If you are playing in any place smaller than that (where there is no need for an engineer) – just watch your gain levels on your tracks, use your mixer EQ sensitively and you’ll be fine.
using technology to improve our craft is great but we are getting to a point where there is a limited return but a huge additional burden of complexity to manage.
these recommendations really are overkill. If you are playing a place where consideration of the room really is an issue then there will be in an in house sound engineer who will manage EQ and compression for you as the last part of the signal chain. If you are playing in any place smaller than that (where there is no need for an engineer) – just watch your gain levels on your tracks, use your mixer EQ sensitively and you’ll be fine.
using technology to improve our craft is great but we are getting to a point where there is a limited return but a huge additional burden of complexity to manage.
I handle all the PA for my Party Crew. Sometimes Ill have 10 Djs on a line up.. they might change every twenty minutes. So the sound variety and the input types change very quickly. Even with every DJ running some kind of rediculous re-routing out of traktor into various filters and compressors and mastering chains .. thier sound cards might differ in out put volume so much that it would be completely meaningless for them to go through the trouble and possible system crashes that might occur during the show which is most importantly …. pleasing the audience and not burning thier ears out. Ive been working on a signal chain for a while that goes like this … DBX 386a tube pre-amp and digital converter. All signals come out of a top nothch xone 92 — our main DJ mixer and go into the tube PRE … there it gets the boost and warmth it might need and alot of the differant types of signal get a nice tube brushing. I have a side chain of EQaulization and two very simple mono compressors .. dbx 160a that give the fast attack and release limiting that I need to protect my equipment and keep the signal from overloading the digital conversion that takes place after it re-enters the 386. The 386 uses some shaping algorythims and a sort of digital overload compression as it converts the signal … fattening up the signal rather than creating digital distortion.. im not sure exactly what it is .. but it doesn’t sound like crap when its too loud… from the 386 it travels via aes ebu cable at 96 khz or 44.1 into my motu 896 mk3 where it gets converted and possibly recorded if necessary. also the motu handles the clocking for the 386 — very few hiccups. From the motu it travels again by aes cable into a sabine 8808 system controller where it is split up into the various signals that supply my speaker system. This isn’t very complicated and it really puts a mastered but not flattened sound on the party crew. We can record a demo straight off the xone that doesn’t get all crazey on the levels and it makes alot of people really happy to hear thier set all under-control — when they could only hear thier own volume changes through the monitors. Monitors are direct feeds off the 92.
TORQ is the winner here!!!!
How would you route this just using the built in mac soundcard? Whenever I try to do it and play the songs I just get this nasty noise everytime i play a track
That’s probably your sound card not keeping up. If you’re doing internal mixing, you should expect to spend $3-500 on ebay for an audio interface that can give you low latency, quality audio. I go from traktor into ableton, and then I A/B out to my 2 channel dj mixer, and back in to a return track.
I’m doing that so I can a) use my silky smooth faders, and b) my mixer has a really good send/return loop to go out to my KP3. By going into ableton I can also mix in me playing a drum kit or instrument in kore, and whatever else I want. With kore you can really pump up a song by grabbing a synth and playing a melody on top of it. It helps to glue the tracks together too when you’re fading between songs, because you can keep a rhythm, and morph it on the fly. I’m experimenting with live sampling 4 bars in maschine and then mapping it to pads. It’s a cool way to chop up loops live. Ableton lets me have eq with sweepable mids for each channel, compression, and whatever else I want. So I save ~$4000 not having to buy a good mixer and rack units. But the only reason I can do that is I spent $600 on a sound card that will let me pump sound through it super quick. For me anyway, more than 7ms of lag messes up my sense of timing.
That’s probably your sound card not keeping up. If you’re doing internal mixing, you should expect to spend $3-500 on ebay for an audio interface that can give you low latency, quality audio. I go from traktor into ableton, and then I A/B out to my 2 channel dj mixer, and back in to a return track.
I’m doing that so I can a) use my silky smooth faders, and b) my mixer has a really good send/return loop to go out to my KP3. By going into ableton I can also mix in me playing a drum kit or instrument in kore, and whatever else I want. With kore you can really pump up a song by grabbing a synth and playing a melody on top of it. It helps to glue the tracks together too when you’re fading between songs, because you can keep a rhythm, and morph it on the fly. I’m experimenting with live sampling 4 bars in maschine and then mapping it to pads. It’s a cool way to chop up loops live. Ableton lets me have eq with sweepable mids for each channel, compression, and whatever else I want. So I save ~$4000 not having to buy a good mixer and rack units. But the only reason I can do that is I spent $600 on a sound card that will let me pump sound through it super quick. For me anyway, more than 7ms of lag messes up my sense of timing.
I use Platinum Notes software to remaster all my music through a series of similar filters into WAV. It seems to do most of this (not the vintage effects of course) is that a possible solution as well?
Good article… One caveat though: While using Jack to route, you cannot use timecode in TSP2 without removing sound card checks in the application itself. Unfortunately reverse-engineering is required, and I’ve fought tooth and nail with a few NI-employed mods on the DJTT forums to get this fixed, and nobody’s interested. Instead, legit users that want to route while using timecode had better learn x86 asm (and reposition their moral compass). 🙁
You can use timecode using jackrouter. I use them in my configuration.
Hey Chris, good article. Glad to hear mention of Jack – I love it, and it’s an essential part of my rig. I’ve never had any issues with it, and I think it’s much more flexible than Soundflower. As for PSP VintageWarmer, I find that introducing that into the signal chain creates unacceptable latency for djing – have you noticed that? Great for studio work though. For a low latency warm-up, I like to use a BBE D82 vst, and Ableton’s native eq, saturator, opto compressor (very light settings) and a limiter, in that order. I set the levels so that the limiter almost never engages. Works well to fatten and smooth.
Hi Zac – the ‘full fat’ vintage warmer definitely adds a bit much but the LE is more subtle, I feel there’s a ‘your mileage may vary’ moral to the story with this method in general because one’s style dictates how much latency you can put up with, to a certain extent. Thanks for the tips I look forward to giving them a go!
Sounds like terrible advice, most music nowadays is already squashed lifeless with compression, limiting and distortion. Adding even more on top is not a good idea. What you’d want to do is matching the RMS values of your tracks, that’ll get you fairly close an even and nice sounding mix. Besides there’s already a limiter on traktors output, and it’s set fairly aggressive already. Unless you’re playing unmastered material, you are very likely to do more harm than good.
That’s definitely a point people need to consider. Mixing via proper gain structuring is the first skill djs should learn. Overuse of compression on already super-compressed tracks will make things sound bad, fast. I think the key is to really learn compression inside and out. I use a compressor on my master out, but I set everything to be really light, smooth. No hard knees, small ratios. It only kicks in during loud mixed sections. It IS a trade off, but if you do it right, you won’t wreck your sound. You do add a touch more compression to already compressed tracks (only at mix points), but you benefit by smoothing out dense mixes.
gosh, i wish i could turn of traktor’s compression. i hate it
you can do in: settings > mixer > set limiter off > set headroom to 0 db … but definitely take care that your signals are not clipping in the channels and in the final mix. mix at low levels and make it louder after mix is ok
I thought Traktor has a limiter on the master out already? Why do you need another one?
Nice article Chris.
I’ve also not had major luck with Jack or even Virtual Audio Cable. Both seem to cause issues with the sound and Traktor (with S4). It could be how I set it up though or the hardware, so I wouldn’t say it is the software at this point. This article makes me want to keep trying though. Thanks!
scamo
Nice article Chris.
I’ve also not had major luck with Jack or even Virtual Audio Cable. Both seem to cause issues with the sound and Traktor (with S4). It could be how I set it up though or the hardware, so I wouldn’t say it is the software at this point. This article makes me want to keep trying though. Thanks!
scamo
Nice article Chris.
I’ve also not had major luck with Jack or even Virtual Audio Cable. Both seem to cause issues with the sound and Traktor (with S4). It could be how I set it up though or the hardware, so I wouldn’t say it is the software at this point. This article makes me want to keep trying though. Thanks!
scamo
I am personally not super-keen on Jack as it tends to have a bit of Lag & crashes my system from time to time. I would propably opt for using hardware EQ & Compressor added onto the end of the chain. Kerry Chandler & many more use it & seems to be a dope way to keep the sound consistent on using old & new trax.
When using mp3 I put all the tracks through an app called switch that has a normalise function available. that helps pretty much just to cover the bases.
i tried jack with Babya AU Host plus PSP Vintage Warmer plus Traktor and i cant get the audio get through the host software. any tips?
i tried jack with Babya AU Host plus PSP Vintage Warmer plus Traktor and i cant get the audio get through the host software. any tips?
i tried jack with Babya AU Host plus PSP Vintage Warmer plus Traktor and i cant get the audio get through the host software. any tips?
Yep – ignore my brain fart on this one 😉 Go with Rax or Reaper. Rax is more dedicated to this task, Reaper is more fully featured and won’t expire on you in 15 days…
Yep – ignore my brain fart on this one 😉 Go with Rax or Reaper. Rax is more dedicated to this task, Reaper is more fully featured and won’t expire on you in 15 days…
Brand new site! Just saw it right now, and you still got me DJTT. Good article, but for an amateur like me, I miss some praticle examples. Anyway, you guys keep me going back every week. Thanks a lot!
Greetings from Brazil (hey Ean, you should come to play someday)
One word for those interested in a kick ass ultra low latency AU host: AULab. It’s part of the develloper tools of OSX. Used to be free, version didn’t change with Lion. You can use the app independently of Xcode being installed. Find a friend that has the devtools installed, zip the app and grab the archive..
Hey, I didn’t know Jack was ported to Windows. I have been using it for
years on my Linux machine. You could, by the way, run a Linux Live
system (no install required, runs from CD/DVD or USB-stick) on a second
computer as a real time sound processor. Does not need to be the newest
one, just a decent audio card. (PCI cards with 24bit/96kHz are dirt
cheap now). There are even specialized Loonix live distributions for
that purpose, like dyne::bolic. As they are all free as in “free beer”
there’s no risk in trying. Worst case you lose some time. best case you
get some independent recording/mastering hardware for scraps.
it’s great you guys are mentioning JACK, which is one of the coolest audio things out there for Linux. For those brave enough to try mixing in Linux, I belive there are some VST hosts for JACK, like fvst, which use WINE. Of course setting this up is hardly an easy task.