For today's free and easy music producers, the knotty business of music theory has always had a slight whiff of the uncool about it.

Talk of Major and Minor scales conjure up images of dry, dusty, bewigged baroque musicians from centuries past, the nightmare of childhood piano lessons, or baffling overheard discussions in jazz clubs concerning the superiority of augmented 13ths over Neapolitan sixths.

And don't get us started on the joys of soloing in the Mixolydian mode…

This bad rep is entirely unjustified, however, since, regardless of whether you're a composer, songwriter, producer or DJ, shoring up your musical knowledge with a smattering of grassroots theory can have a profound effect on the quality of your output.

Here at MusicRadar, we never underestimate the importance of a good grounding in music theory. After all, how music works is the key to everything we do when we're creating a project.

So instead, try to think of theory as a framework to support and inspire your musical endeavours, rather than inhibit and impose rules on them.

The best part is that you don't have to plumb the murky depths of the subject to be able to use it to your advantage. When a crippling loss of creativity strikes when you're sitting in front of your DAW – and this can happen to any of us at any time, let's face it – having a few tried, tested and theoretically proven techniques in your back pocket can be a lifesaver.

Here, we're going to highlight ten of our favourite, most practical theory-based tips that can help to kickstart the creative process next time your inspiration well runs a little dry.

Let's knuckle down and get theory-ous!

• Best online piano lessons: recommended piano lesson apps, software and websites

1. Broken chords

Should your keyboard parts naturally tend towards the clunky side, things can be improved by employing a technique known as broken chords.

This involves breaking chords up into their component notes and playing them as short arpeggios (runs of notes). The technique doesn’t just add rhythmic interest, but it can be a great way of coming up with ideas for cool melodies that work naturally with the chord progression, since the notes you pick out are taken from the chords themselves.

Here we have a four-bar progression of simple block chords. To the right, we have the same progression, but played using broken chords. Like finger-picking on an acoustic guitar, instead of playing all the notes on the same beat, each triad is split into individual notes.

2. Animating chords with inversions

Moving the order of notes around within a chord can take a progression in new directions without changing the actual chords. This is easily done by selecting notes and shifting them up and down an octave - most DAWs have a shortcut key for this, such as Shift-Up/Down.

Here we’ve taken a simple C, D, Eb, F progression and completely altered its shape by swapping the note order around in the second half. These ‘rearranged’ chords are known as inversions.

3. Change the bass note

The notes a bass part plays can assume the role of root note for the chords played by parts above it. Sticking different bass notes underneath can radically alter the sound of a static chord sequence.

Step 1: Let’s illustrate by looking at a piano part playing a repeated rhythmic C minor chord sequence over and over… and over. It sounds pretty boring, as the chord doesn’t change. Instead of thinking up new chords, though, let’s try injecting freshness by using a more animated bass part.

Step 2: For the first two bars, we stick to a C bass note, as it’s the root note of the chord. After two bars, however, we shift the bass note down from C to A, forming a completely different chord with the piano - an Am7b5 (A-Eb-G-C), in fact.

Step 3: Keeping the piano chord the same, we next move the bass from A to Ab, making an Abmaj7 (Ab-Eb-G-C) chord. Then down to F, up to G and back to C again. If you’re not sure what bass notes will fit, try picking notes that are already in the accompanying chord(s).

4. New flavours with modes

If you’ve never encountered modes before, this one tip could change your musical outlook forever. Scales are a basic component of music theory - most people know one or two of the major scales, the most common being C major (C D E F G A B). But what if we told you that you could unlock a secret world of evocative alternative scales simply by playing any scale from a different starting note?

Try it by playing C major in the conventional fashion, from C to C on a MIDI keyboard. Now give your ears a few seconds’ rest to forget the sound of it, and play the same scale again, but without starting on C. Try going from D to D. Or F to F. Notice how they have a different quality to the major’s happy, kindergarten sound? What you’re playing are different modes of the C major scale, and they sound different because of the different sequence of intervals between the notes that happens when you start somewhere other than the root note of the scale. Underpinning each new mode with a bass note the same as your new starting note will reinforce the effect.

5. Monotony rules

Melody-writing can be tough, but sometimes, if the rhythm is hooky enough, sticking religiously to one note is all you need to do.

Try using a monotone melody in a verse part, and slowly add small variations through the bridge section to build to a chorus hook. This approach also gives you an opportunity to focus on rhythm and sonic variety, rather than just melody alone, for adding interest.

6. Borrowed chords

Diatonic means “in key”, and diatonic chords are formed using only notes taken from a particular scale. So a C major scale containing the notes C D E F G A B - the white notes on a keyboard - would give you the diatonic chords C, Dm, Em, F, G, Am, and Bdim, formed by stacking alternate notes onto each note within the scale. These are the basic triads you may already be familiar with; however, a palette of only seven chords can be limiting, so why not borrow chords from other keys?

Borrowed chords are most often taken from parallel keys - keys that have the same root note as the original key. In the case of C major, the parallel minor key is C minor, so we have a whole new set of chords to choose from - Cm, Ddim, Eb, Fm, Gm, Ab and Bb.

7. Layer chords together

For an exotic sound, try combining regular major or minor triads together to form polychords.

For a classic polychord combo, let’s use C major as the first chord (C-E-G). For the second, we need to find the major chord that has its root a perfect fifth above the root of the first chord. In this case, that happens to be G major (G-B-D). Play the two chords together and you get a Cmaj9 chord (C-E-G-B-D). If you’re stuck for which chords to combine, go for ones that already have a note or two in common.

8. Take some giant leaps

To give things a kick when putting together a melody, try making a dramatic leap up the keyboard. A good interval to use for this is a minor or major sixth.

You find these intervals by using the scale degree numbering system outlined in Tip 10. With major scales, the distance between the root note and degree 6 of the scale is a major sixth, so in the case of C, a major sixth above would be A. Jump up from C to A in the course of a melody, and it’ll take you up into a whole different area of the keyboard.

9. Build an extension

Extending regular major or minor triads by adding extra notes can radically transform a track. For instance, in C major, a Cmaj7 is made by adding the seventh degree of the C major scale (B) to a C major triad (C-E-G).

If we extend the scale up the keyboard beyond the octave, we get into the extended range, where a ninth is essentially the scale’s second degree played up an octave. So an easy way to play a ninth chord is to play a root-position triad (for example, C major – C-E-G) and add in the second degree between the root and the third (C9 – C-D-E-G).

10. Go pentatonic

Limiting yourself to notes from a pentatonic scale may give you a smaller choice, but it can give rise to some memorable melodies.

Step 1: While a conventional major or minor scale contains seven notes, a pentatonic scale only contains five. Effectively, they’re regular scales with a couple of notes removed. If we look at a normal C major scale, we can see that it contains the notes C, D, E, F, G, A and B.

Step 2: We can then give these degrees numbers, from 1 to 7. To make the scale pentatonic, we need to remove degrees 4 and 7 - in this case, that’s F and B, leaving us with C, D, E, G and A, the C major pentatonic scale.

Step 3: For a minor pentatonic scale, omit the second (D) and sixth (Ab) degrees. For the C minor scale, we’re left with C, Eb, F, G and Bb. Pentatonic scales sound great, and they’re fun and easy to use.

While a mixing console may initially seem like an impenetrable maze of buttons, sliders and knobs, it’s actually a very simple to understand ‘what you see is what you get’ machine when you get down to it. And we're here to help.

The principles of using a mixing console in real life versus an on-screen in-your-DAW alternative are basically identical. That’s because your DAW essentially copied the tried and tested layout of the classic physical mixing console that’s been the focal point of recording studios for over 90 years.

After all, if it ain’t broke, don’t fix it.

Tracking versus mixing

Getting right back to basics, a mixing desk essentially operates in two different and distinct modes. First of all comes the tracking, where all sounds are incoming and your desk is controlling the levels reaching your DAW (or old skool tape) and the headphones and monitors of the band in the studio.

And if you’re mixing a live performance exactly the same applies. That is, sound comes in, is tweaked to perfection, then heads out again.

Once the performance is over and captured perfectly in your chosen recording method, your mixing desk is then used for its second dual purpose – mixing. Now, rather than recording your band, your mixer is listening to what was recorded. Your band can head home and you can get on with the important business of making them sound great in their absence, with each separate performance appearing on a separate channel of the desk.

Big or small… They’re all the same

The trick to viewing and understanding a mixing desk is to see it as a set of individual channels all lined up side by side. Commonly mixing consoles (and let's just call them ‘mixers’ or ‘desks’ for short) will feature eight or 12 such channels. Larger home studio mixers will go to 16 or even 24. While in a professional studio it’s not uncommon to see up to 96 channels side by side.

But no matter what the size, if you look closely you’ll see each separate channel strip, running vertically from the fader at the front surface of the console, all the way through to the outputs on the back, all identical, all lined up, side by side across the desk.

Each of these channels is available for use by an instrument, be that a vocalist, guitarist, snare drum and so on. And each channel has an input, a volume slider, some controls to tweak the sound and an output. And that’s it.

Each channel’s output is combined into the ‘master output’ and that heads out to your amplifier and speakers with a final volume slider setting the listening level.

In short, the aim is that all the elements of your track, appear on the mixing console and the engineer or producer at the controls can tweak each channel to make the combined output sound perfect.

And, if you understand the workings of a single channel then you’ve got the basics of the entire desk.

What’s in a channel strip?

Starting from the front edge of the console and working through to the back here’s what you’ll most likely see:

First of all there’s a fader – a long, smooth-sliding slider – that’s used for adjusting the volume level of the instrument that’s plugged into it. You’ll use this to set the level of the instrument coming in. Be that while you’re recording – making sure the level is loud but not too loud – and then how much you can hear when you listen back – be that for mixing down a multi-track recording session for a finished mix, or getting the sound right for a live performance.

When recording you want all the sounds recorded at clear, workable volume levels and the mixing console is essential for bringing unruly loud instruments (such as drums) into line with softly spoken vocals or percussion, for example.

Smaller desks (and some classic older units) will feature a knob instead of a fader.

Above this fader usually comes the pan knob. This adjusts the balance between the left and right speakers for that particular sound. Thus you can position a sound within the stereo spread all the way from hard left, to hard right, with dead centre on the knob being an exact 50/50 split between the two speakers.

This allows you to place sounds in space and stop ‘rival’ instruments (say hi-hats versus shakers) from clashing in the frequency spectrum. Placing one to the left and one to the right gives each room to breathe and allows the listener to hear each clearly and enjoy both.

Usually near the pan you’ll find mute and solo buttons. Hitting mute will turn that channel off, allowing you to silence the instrument on it without having to zero the (expertly positioned) level of the fader. Alternatively, hitting solo mutes all the other channels, allowing you to hear that particular channel on its own. Essential for zooming your ear in, to make sure each instrument is perfect.

Next – and occupying a large centre section of each strip – comes the EQ which lets you adjust the bass and treble (and on bigger consoles, everything in between) for each sound.

Smaller consoles will have a basic low and high EQ. A ‘low’ knob will allow you to cut or boost the bass frequencies of the sound on that channel, while a ‘high’ knob will allow you to do the same for the treble.

On larger consoles you’ll find – in addition to the low and high knobs – a third ‘mid’ frequency range, often with an adjustable ‘sweep’ knob to allow you to more precisely pick out which of the mid frequencies you’re cutting or boosting.

A ‘sweepable mid’ is perfect for adding detail to important sounds such as a lead vocal or lead guitar line, ensuring that they ‘cut through’ in between the bass sounds of the kick drum and bass guitar and high frequency fizz of the hi-hats and percussion.

The bigger (and more expensive) the console the more EQ controls you’ll get. Top end desks will feature multiple sweepable bands (in addition to those low and high frequency filter controls) allowing more precise cutting and boosting of the sound on each channel at multiple frequencies simultaneously.

Certain mixing desks are famous for the nature of their EQs, injecting distinctive warmth to the sound or highly regarded for their ability to pull disparate instruments together as a solid, digital ‘block’.

Meanwhile, round the back…

By now we’re getting towards the back of the channel strip where you’ll find a few of the more technical, workman-like controls that, once set for the instrument, are most likely never touched again during your recording session and mix.

Right at the back edge of the console you’ll find each channel’s inputs – most usually in a variety of sizes and shapes – allowing you to plug in whatever you need to add. There’s controls for the channel's input level or gain, allowing you to control how ‘hot’ the incoming signal is, so that that particular sound can sit well on the desk and alongside the other instruments in your mix.

And – as with the EQ – on famous and classic desks, it’s often this input pre-amp that’s responsible for a certain desk’s character or warmth, meaning that certain desks (and their virtual plugin emulations) are highly sought after thanks to their unique pre-amps on each channel.

Adjusting this gain control allows the quietest sound (a simple shaker, for example) to sit alongside the loudest (a blistering solo from a Marshall stack) and for all your track’s elements to then be mixed together as equals.

Watch out for a Pad knob too, which will reduce a signal level to prevent unpleasant clipping (aka distortion).

High end extras…

We’ve already covered most of the basics that you’ll find on every desk, which just leaves the ‘special’ controls that you’ll find on high end consoles (and their virtual DAW equivalents).

Some desks will have built-in de-essers or noise gates, allowing you to fine tune spikes in the levels of the recorded performance, and while separate ‘outboard’ boxes will, of course, do the same job it’s clearly more convenient to have such features built into the channel strip of the desk.

And such per-channel compressors and desks from the likes of Solid State Logic (SSL) made a great play of the quality of their per-channel, built-in compressors (in addition to their ease of use and simply having them ‘there’ for any sound that you have on a fader) with their sound becoming part of their desk’s mystique and going some way to justify their impressive price tag.

Likewise the SSL’s (and other brands’) master bus compressor. Large desks will have a further final mastering compressor on the desks final stereo output, allowing you to give the combined might of your finished mix just one final cohesive polish on its way out to your amplifier and monitors.

Watch out for high-pass filters per channel too. These can be used to quickly remove low end rumble from mics on a hard studio floor for example, or unwanted ‘ghost’ bottom end sounding from badly positioned kick drums and bass amps.

More professional desks will also feature buttons such as Phase Reverse, allowing you to flip left and right channels – ideal if you’re working with multiple sets of stereo left and right mics (such as over a drumkit). Overlaying a pair (or group) of mics listening to the same sound can produce strange phasing effects that can sap their power and clarity. A quick press of Phase Reverse will pop everything back to ‘normal’ again.

You’ll find dedicated sliders for separate monitor mixes too, allowing you to deliver bespoke mixes to each band member’s headphones. The vocalist will always want themselves nice and loud with plenty of effects already added, while your drummer would much rather be listening to your bass player.

And, what was once a luxury on only the most expensive desks, is now a feature you’ll find in any DAW – the ability to record and replay fader and knob movements. This ‘automation’ transformed music in the late 80’s, allowing producers to really perform at the mixing desk and for music to become more complex with chops and changes and sounds and effects suddenly appearing and disappearing.

Today, while only larger professional physical mixing desks feature such ‘flying fader’ automation, every computer-based DAW will let you record knob tweaks and level changes as you mix, freeing up your hands and mind for further experimentation elsewhere.

Finally, special mention to that hardware studio stalwart, the Talkback button – a luxury of only the largest desks. Pressing Talkback enables the desk’s microphone, allowing the producer to be heard over the band’s headphones and monitors. Perfect for delivering a little ‘can we try that one again?’ encouragement to the band…

Take the bus…

So now we’ve got all our instruments – be they coming live from your band or being fed from your recording – lined up on your desk and – thanks to the gain controls on the inputs, EQ, and volume level fader – sounding at least ‘something’ like you want.

Next comes a little bit of mixing desk magic.

On larger home consoles (and on every professional desk) you’ll find mix buses. These allow you to route groups of instruments together and mix them as one. So rather than sending each of your drum channels out to your left and right speakers as usual, you could instead send them to ‘Auxiliary Channel 1’ or ‘Mix Bus 1’.

Give this ‘Mix Bus 1’ its own channel – often labelled as a Sub Group on larger desks – and you can then control multiple instruments on a single fader.

Doing so not only simplifies your mix – want to turn down ALL the guitars? Simple, they’re all on a single fader now – but by grouping tracks together you can process them as one. Also – via a little mixing desk magic – grouping together ‘likeminded’ instruments on a bus allows them to stew a little in each other’s juice and often leads to a pleasing, more cohesive sound.

It’s a quality of classic mixing desks (and one expertly recreated in their software emulations) that sending all your drums to a ‘Drum Bus’ and then perhaps using a compressor on this single, multi-element, channel somehow ‘glues together’ their sound.

Why not send your bass to the same bus too, to share a little bit of that compression magic?

Suddenly, rather than your track sounding like disparate elements all fighting for air, you’ll have a tight, locked down rhythm section all working together and all on a single fader.

Effective effects

As mentioned earlier, some larger desks will have compressors and de-essers built in (and owners pay handsomely for such features) but you don’t have to simply use what the desk has to offer.

Each channel – even on small mixers – will feature at least one Send control. Often two or three. These knobs will send a set amount of that channel’s output out to an Auxiliary mix bus (a dedicated bus set aside for such things). Thus, place a reverb effect on Effects Send 1 and you’ve got a reverb that can be used by every channel on the desk.

Turn up the send on your vocal channel to swamp it in luscious reverb. Add a little reverb to your snare channel. Get your synth pads sounding ‘big’ by nudging up their reverb send too.

And why not place a chorus effect on Effects Send 2? Send most of your acoustic guitars there… And maybe some of your backing vocals too.

Such effects buses are a great way to quickly inject some sparkle and magic into your mix. And they let your effects boxes go a long way – no need for an effects box for every channel when they can all share one.

Of course, this means that you’r using the same reverb setting or delay time for your entire mix, so it’s always advantageous to have three, four or more effects to choose from, each on a different Send. And – thanks to the modern DAW – if your computer can handle the processing (and these days they always can) you can have separate instances of your favourite effects on every channel. EACH with their own settings and levels.

Thus, the simple shared ‘Effects Bus’ and ‘Send’ is becoming something of a has-been… But it’s a good principle to learn and understand, as limiting yourself to ‘one reverb and one delay’ old-skool style is a great way to learn mixing, avoid getting confused by hundreds of options, and allow your mix to ‘sound great’ fast.

Perhaps use a simple ‘effects send’ mix set-up to get your track into basic shape – a shared reverb, a delay and a chorus. Then create more bespoke per-channel effects chains later when you’re ready for the finishing touches?

The sum of its parts

Hopefully by now you’ve got the measure of the desk in front of you. Just remember that all desks, small to big, from four channels up to 96 are essentially the same. Don’t get blinded by science.

Focus on the parts you need – level, pan, EQ – to begin with. Then add compression, use effects, group buses, as and when you need.

Keep it simple… Then keep experimenting.

PLUGIN WEEK 2026: The very term ‘saturation’ sometimes feels like it has a negative value attached to it. Throughout recording history, technology has sought to deliver the cleanest signal possible, so why for the love of Joe Meek would you want to turn a clean signal into a distorted one?

It turns out, the answer is relatively simple. As human beings living in an organic world, we quite like impurity and a measure of chaos - and that's where sensible use of saturation can bring an awful lot to the musical table, whether it's individual instrumentation, as well as an entire mix.

The term distortion is very closely linked to saturation, but while they are subtly different, one could not have existed without the other.

If we go back around 100 years, the predominant music of the era was jazz, and particularly big bands. Electric guitar amplification was not particularly advanced, and would now be probably described as ‘lo-fi’.

Guitarists would often turn up their equipment as high as they could go (we don’t think ’11’ was a thing back then), in an attempt to compete with the loud acoustics of a large band. This would lead to equipment overdriving or speaker cones failing, with the resulting sound being something close to the guitar distortion that we know today.

With the advance of electronics, distortion pedals began appearing in 1962, and by the 70s, they were both commonplace and an essential part of a guitarist’s rig. This gave guitarists ultimate control over their distorted sound, without the need to destroy an amplifier on a weekly basis!

But why should guitarists have all the fun? With the rise of synthesisers in the 70s and 80s, synthesists and producers realised that they could get some pretty gnarly tones by attaching the same distortion pedals to synthesizers.

Nothing exemplifies this more than the acid house era of dance music, from the late 80s and early 90s.

It was this period where the now-legendary Roland TB-303 was repurposed from a humble bassline machine to the must-have device for the genre. It turns out, whacking up the resonance on the 303 filter produced an amazing sound through a distortion pedal. Club-avoiding guitarists, however, were largely non-plussed.

So what exactly is saturation, and what does it bring to the production table?

By and large, saturation is designed to be more subtle than distortion, often with a penchant for mimicking good old-fashioned analog equipment. You remember our previous point about clean signals? We rest our case!

Saturation excites and distorts certain frequencies and harmonics in the signal, depending on your chosen plugin. What's more, we can trace the saturation story back to the late 90s.

Back then, CDs were the predominant format, and many musicians and producers were reporting that the overall sound of the digital signal chain was a little cold and sterile.

Couple this with the emergence of one of the first rack-mounted units to offer a one-stop mastering solution - the TC Electronics Finalizer - and basic mastering saturation became commonplace.

TC Electronics described this saturation as 'Digital Radiance Generation', or DRG for short.

DRG applied second harmonic distortion to a mix, designed to mimic analog equipment and specifically analog tape, so it's no surprise that it was quite a hit.

However, over-using it could quickly destroy a mix, forcing mix engineers to pay close attention.

Fast forward to the current DAW age, and we are now flush with a plethora of different saturation plugins, with many being very highly regarded and offering bespoke solutions, depending on your production requirements.

One of the big boons of plugins, is that you can use them as many times as you like, and while it is probably a bad idea to sprinkle saturation on every instrument in your mix, there are definitely places where employment can pay dividends.

If you find yourself recording an electric bass part directly via your audio interface, you could choose to use amplifier simulators to colour the signal. This could be regarded as a form of saturation, but you could also go all-in and just apply a saturation plugin to warm up the signal.

One of the most highly regarded in the current market is Decapitator from Soundtoys. Not only is it flexible, but it can go from incredibly subtle to total extreme very quickly.

When applying saturation, it is always really important to listen objectively. The moment you feel the signal is getting too aggressive, it's time to back off, and that may not necessarily be exclusively down to the saturation amount control.

Just about all plugins will be equipped with a Wet/Dry see-saw control, and adjusting this can scale an extremely saturated sound to help it blend in more nicely.

Of course, this principle could also apply to sampled bass sounds, which may be a little too clean for your mix. It's also a concept which could be applied to drums, specifically if you are working with samples within your DAW.

It's a common complaint that many drum samples are just too clean, so it could be very useful to add a little saturated colour to your drum track. The only problem is, the drums themselves often sound great with saturation, but the hi-hat and cymbals can react less favourably…

A sensible solution requires the creation of 2 drum tracks, both with accompanying MIDI parts. You can then mute all of the drums on one channel, muting all of the metallic instruments on the other. This leaves the path clear to saturating the drum elements on their own, with no effect on the cymbals and hats.

Before you get lost in instrument saturation heaven (or hell, if that's your preference) be aware that some instruments do not embrace saturation so readily, particularly instruments with a more acoustic persuasion, such as acoustic pianos, string sections or brass and woodwinds. There's no harm in trying, but don't be surprised if saturation does little to improve the sound.

With a slight sense of history repeating itself, we have to revert to TC’s DRG concept, and discuss how beneficial subtle saturation can be to an entire mix.

Mastering suites, such as iZotope’s Ozone, offer a number of saturation solutions, designed to be used during the mastering phase. These include tools to excite upper frequencies, and elements for inducing the sound of vintage tape.

But if you want a little more control, one company that personifies the application of saturation to mimic analog is Universal Audio. It has produced an array of exemplary plugins which are perfect for subtle saturation across your mix.

If you are nervous about over-applying saturation, take a look at the Verve Analog Machines plugin, which is available in two forms; the first is an entry-level Essentials version, offering four saturation models, while the upper-level version provides 10 models.

The idea is simple - apply across your mix, and adjust the tone and amount controls to suit your production. The included models range drastically from subtle, through to downright dirty, so it's important to be objective.

For even more control, UA’s emulation of the Ampex ATR-102 Mastering Tape Recorder is stunning. This plugin can induce everything from saturation through to hiss and even wow and flutter. It just depends how extreme you want to go!

One final thought. We've mentioned on several occasions that you need to be objective about your saturation exploits, and be careful not to overuse. You can easily destroy a mix by being a little overzealous.

If you find yourself regularly creating or delivering stems of your mix, be aware that many saturation plugins induce noise.

If you deliver half a dozen stems and they all have noise on them, when all the stems are played back together, that's going to be quite a racket. It’s probably best to defeat the noise element as much as you can, in this setting, so that your music doesn't get drowned out by hiss.

We spent many years trying to get rid of tape hiss and noise, so you don't want to end up back where we all started.

PLUGIN WEEK 2026: Reverb is a hugely versatile tool that can be used to create all sorts of effects and transitions. Here, we're going to show you how to use reverb to create filters, reverses, risers and pads...

Let’s start by creating some interesting filter effects. Import a drum loop or other sound source into a new DAW project, insert an algorithmic reverb, and set its wet/dry balance to 50%. Choice of reverb has a big impact on both the method and results here, so you may need to experiment to get the effect to work.

What we are essentially trying to do is create a controlled feedback situation. A small room/reverb size combined with longer decay times tends to do it, and some processors also have a feedback setting which does the trick nicely!

Pre-delay can have an influence here, too. In short, experiment with parameters and plugins, and you’ll find a whole bunch of interesting filter effects.

Let’s make a real reverse reverb now. Create a new project and import a vocal or other lead part. Insert a reverb on the vocal and set up a pleasing, medium-to-long effect. When happy, use your DAW’s audio processing tools to reverse the reverb part – not all DAWs can do this, so use a free audio editor such as Audacity if needs be.

Import the bounced audio to a new track if your DAW didn’t do this automatically, then reverse it once again (or, if using Audacity for processing, reverse and then import to your DAW). Carefully position the reversed bounced part so that it lines up properly with the unreversed original part, then mute that dry vocal’s track. You now have true reverse reverb!

Risers use similar techniques to reverse reverb, but tend to be less of a faff. Open a project and find a point in the song that would benefit from being preceded by a riser. Export a short segment of audio from this point – sufficient to seed the reverb, and including any tracks that you want to include in the riser sound.

Create a new project and import the snippet you just exported. Insert a reverb, set it to 100% wet, and create a sound with a long tail and plenty of character. Add other plugins for more sonic interest if needed. When done, export the results, import this new file back to the original project, reverse it and position it.

Let’s now create a synth pad. Open a new project and create a new instrument track using a synth or sampler that supports granular playback. We’re going to use Arturia Pigments, but you could use NI Kontakt, Steinberg Padshop, BLEASS SampleWiz 2, or any one of numerous premium and free granular synths should you not already have something suitable.

Import into your instrument into the unreversed riser sound you created earlier. Use the synth’s sample and granular editing controls to hone in on a pleasing portion of the source sound. Build up a patch for the sound by adding a pad-like amp envelope, and modulating the granular position time to create timbre shifts.

Finally, let’s play with some freezing. Import a drum loop into a new project and insert a reverb that has a freeze function – we’re going to use BLEASS Reverb. Set up a nice ’verb, add an automation track for the freeze button, and create a rhythmic automation pattern. You could then enhance the effect by automating other parameters.

PLUGIN WEEK 2026: Can you can remember exactly where you were when you heard your favourite album (or a life-changing track) for the very first time? We are fortunate enough to have had many moments like that. But there was one particular album where this writer can remember calmly sitting down in front of a (now old-fashioned) hi-fi system, and having his mind thoroughly blown from the very first track onwards…

I think we can all agree that Radiohead are quite ‘the’ experimental band. Their exploration of the various extremes of music have taken them to the very fringes of what’s usually acceptable in the studio - including the use of some exceedingly cutting-edge production techniques. They’ve also incorporated a fair share of vintage instrumentation, such as the Ondes Martenot, as well as strings and well… you name it, Thom, Jonny, Ed and co have probably done it at some point.

Having experienced exceptional success with their more chart-friendly albums The Bends and OK Computer, the creativity dial was set to maximum for the electronic-leaning follow-up, Kid A in 2000.

The crystalline simplicity of the opening track, Everything in its Right Place (which we’ve analysed at length in an exhaustive song story here), was the first indication that we were in for something special…

The use of Prophet-5 synth as an accompaniment to Thom Yorke’s audio-scrubbed vocals (a relatively new technique which was made possible by Pro Tools-based production) set the tone for an album that would introduce millions to sounds and innovative attitudes previously the remit of a select, Warp Records-signed few.

By the time the third track on the album made it to the front of the queue - the dark, circular bass groove and frenzied, blaring horns of The National Anthem - our socks were thoroughly blown off.

One thing that we particularly remember being amazed by at the time was the exceptionally cool effect being applied to Thom's vocals.

“I think, [Thom] is someone who's very into the fine voice that he has, and at the same time he's always looking for new ways to try and express different personae,” said bassist Colin Greenwood back in 2001 in an interview with the BBC. “A way of doing that is by treating the voice”

Although we can’t be fully certain of the chain used to achieve the particular effect, we know for a fact that, at this time, Radiohead were adopters of a largely experimental package which can certainly achieve the same result.

In fact, this particular piece of software formed part of my own regular arsenal of effects prior to Kid A coming out, so I could conceivably stake the claim that I used it before Radiohead!

Just so that we’re completely clear about the glass-like effect that we’re talking about, we have to wait until around 01: 36 on the timestamp of the song, and the first vocal entry - after the main groove and overlaid Ondes Martenot have set the scene.

It is identifiable by the resonant sound that we hear after Thom delivers a line - it seems to leave a lasting impression in the audio, almost the aural equivalent of a bright light being seared into your retina.

It imprints once the vocal line hits a particular frequency. It's an effect that gives a metallic quality to the vocal, and can be achieved using a comb filter.

Casting my our minds back to the late 90s, and computer-based effects had really started to hit their stride, but pre-VST effects and plugins, an organisation called INA GRM (Groupe de Recherches Musicales) had long been established as a leading centre for research into the use of electronics in music, particularly experimental music.

First founded in 1951 by Pierre Schaeffer, it was known as a centre for the development of musique concrète; the groundbreaking technique of creating music from found sounds using analog tape (an early form of sampling, but using tape and razor blades) which gave inspiration to organisations like the BBC Radiophonic Workshop.

In 1990, INA GRM released GRM Tools; an entirely unique standalone suite of software and processing effects, running in a DAW-less capacity on an Apple Mac. It’s where we first heard the GRM Comb and its highly recognisable stylings.

“The first GRM Tools came out in the early 1990s but they didn’t just come out of nowhere,” INA GRM’s director François Bonnet told Ableton. “Actually, some of them were trying to go back to some early machines like the Phonogène and the Morphophone [early innovative instruments]. But mostly they were continuing the first real-time treatment system developed at GRM : SYTER (SYsteme TEmps Reel / Realtime System). The Syter algorithms were really inspiring to start a collection of sound treatments as plugins. So as plugins, they were really quickly designed as commercial software, but they come from a long period of prototypes."

Due to its standalone nature, it meant that many avant garde and experimental composers used GRM Tools with an accompanying recording device, such as a Digital Audio Tape machine (DAT to its friends), recording the experimental output generated by the GRM Tools suite.

With the traction of VST plugins, and the adaptation to other formats such as AAX and AU, GRM moved with the times, offering elements from the original Tools suite in individual plugin form.

We know that producer Nigel Godrich and Thom Yorke were keen users, and so it seems likely that it was the GRM Comb Filter plugin that was responsible for the effect we hear on The National Anthem.

The effect was described more broadly as a ‘ring modulator’ by guitarist Ed O’Brien in an interview with MC2. “We are delay and reverb snobs. I think the interesting thing is, we don't go for nice sounds. For example, we processed the vocals on The National Anthem with a ring modulator. We can get up our own asses a bit, but what we do recognise is, when it comes to sounds, I think we've got quite good taste. We're good at judging what is and what is not a good sound. For us, part of recording is finding out different methodologies and techniques to create new sounds - which is what you should be doing in the recording studio."

It's therefore necessary to have either this plugin (or something very similar) in order to recreate the same effect.

With perfect timing, INA GRM has just re-engineered all of the classic Tools from the original suite into a new standalone and plugin format, which it calls Atelier. Moreover, it is also now available for Mac and PC. Nice!

‘What is comb filtering anyway?’ I hear you ask…

The literal definition is as a form of distortion, which occurs when a single audio signal is combined with a delayed facsimile of itself.

However, in the effect domain, it is quite unlike conventional high or low pass filters. A comb filter offers a number of spikes, a little bit like heavily narrowed ‘Q’ points on an EQ curve.

Also much like an EQ, these spikes are exaggerated and excited by the presence of a matching audio frequency, hence it picks out particular notes and frequencies - which can be defined by the user - to create resonant tones and chordal effects.

In order to achieve a similar effect, you are going to need two things; firstly, some form of audio signal to excite the filter - which we are going to adopt as a vocal part - and some form of comb filter plugin.

You can download a demo version of GRM Atelier if you just want to dip your toe in the comb filtering pool, although the originators are now accompanied by a plethora of other brands that have picked up the comb filtering mantle.

Examples such as Polyverse Supermodal or Kilohearts Comb Filter (which is also free) are all ripe for exploration and can be used to follow this exercise.

Start with a simple key note and build the first frequency plot from here

Once you have your vocal part on a track within your DAW, load up your chosen comb filter plugin.

We need to think about which notes we want to excite from our vocal signal. As many plugins use frequency as their parameter of choice, it's a good idea to nudge the frequency plot up and down, until you find a satisfactory note to excite.

To give you a rough idea of value, the note A above middle C on a piano keyboard, has a frequency of 440 Hz. The note A, one octave higher, has a frequency of 880 Hz - so the rest is pure maths!

Some comb filter plugins also offer a pitch value, so you can simply use a note name instead of a frequency value to construct your desired chord.

In our example, we are using GRM Tools Atelier, but many parameters will be duplicated within other plugins. The Decay parameter adjusts the volume of the resonance, and will dictate the length of time it takes for the resonance to decay to nothing. Be careful, because higher values here can be extreme in volume.

The Damp control limits the number of overtones, so this should be adjusted, particularly if you only want a single note to be exaggerated.

Once you've added your first frequency point, it's time to think about adding a second, third or more, but once again, you need to consider which notes you want to bloom.

If your vocal line or song is in a major key, consider exaggerating the 3rd of the chord (in much the same way that Radiohead did). If you are in a minor key, try exaggerating the 5th of the chord, which will also work well in a major key too.

In our example, our song is in C Major, and we have chosen to place the note C on the top, with a G and D below, creating a Sus2 chord in an inverted state.

A note though, the notes in the chord won’t be constant, only appearing as the vocal line stays across the assigned notes in the plugin, exciting the frequency.

Add additional frequency points to build up your chord

As you add each frequency point, increasing its amplitude will bring out each note.

You may choose to temper and increase/decrease each frequency point, meaning that you can balance the chord to suit your purpose. In all cases though, it is important to listen, as a spirited vocal performance can have an extreme effect on an outputted signal.

Always work with a degree of caution - each time you add a frequency point, and increase its resonance, it will stick out like the proverbial sore thumb.

It's therefore important not to overcook frequency points, which is a sensible mantra for the effect itself.

There is nothing so old and obvious as an overused effect, so it might be worth holding this effect back for a point in the song where it really counts. After all, it took Radiohead over a minute and a half on The National Anthem before Thom Yorke’s comb-filtered vocal made an appearance.

When you get it right - as Radiohead absolutely did - it’s an effect that leaves a lasting impression on the ears of listeners.

PLUGIN WEEK 2026: There’s still something slightly silly - slightly ‘daft’ perhaps we should say - but smile-inducing, about the classic ‘robot’ voice when used on a track.

Everyone from Kraftwerk to Daft Punk and Bruno Mars have explored the concept, but back in the late 1980s, the amazing sound of robot voices infiltrated RnB. It’s a sound that remains ultra-popular and is not too hard to recreate in software, Particularly when we deploy a smattering of choice plugins. But you need to consider some key things first…

The first major question to address, is whether you are actually listening to a vocoder or a talkbox.

Vocoders can be both hardware and software-based and typically rely on audio input from your microphone-based vocal or drum track. By and large, the sound of the vocoder tends to be a little bit softer, often leaning into aligning your voice with prescribed chords.

The talkbox, on the other hand, tends to adopt a harsher and brighter tone, and is visually identifiable by the plastic tube, which is placed in the side of the player’s mouth.

This is relatively essential, but also means that talkboxes can only really exist in the hardware world, and not as plugins.

There are exceptions, of course, with many talkbox plugins on the market, but these tend to be sample-based and pre-scripted, meaning that you cannot apply your own vocal or lyric to your track.

This can be fine if you just want to add the odd ‘Boom’ or ‘Baby’ as a little garnish to your production, but it’d be nice to be more flexible…

You could of course, consider going down the hardware talkbox route. This would mean the purchase of an appropriate pedal, with the MXR M222 Talkbox pedal being the current best and most affordable option.

You’ll then need a synth to provide the tone; the Yamaha DX100 has become the model of choice, but you'll need to source one through the secondhand market, as it hasn't been made since the late 80s. It's also got mini keys, which might not be the best option if you are used to a full-size keyboard.

The other downside with going the hardware route, is the long learning curve. You'll need to practice using it, to make it as comfortable an expressive instrument as possible.

To be frank, this is the tricky bit!

When ‘talk-boxing’, you effectively have to over-annunciate a vocal, without actually speaking or singing!

You also have to do this while gripping a plastic pipe with your molars.

As one professional talkboxer said in interview; ‘The tube is a slippery little sucker!’ This might be why you see so many talkboxers continually adjusting the tube, while desperately trying to deliver a lyric. It's the musical equivalent of ventriloquism, and ironically, saying a phrase like ‘Bottle of Beer’ is just as difficult!

So if you really want that sound, what are your other options?

You could of course hire a professional.

Bruno Mars hired Mr. Talkbox (the clue is in the name) to deliver the amazing sound at the beginning of 24K Magic, although to be fair, Mr. Talkbox is effectively re-creating the sound of the original master, Roger Troutman.

He was the lead member of the band Zapp, back in the 1980s, before sadly passing away in 1999. But, his creative legend lives on.

“I was just a music head. Then one of my cousins introduced me to Roger Troutman and his band, Zapp,” Mr Talkbox said in an interview with New Release Today. “He played a song called Computer Love. I was like, ‘What in the world is that?’ I asked him if there was more of that music. And, he told me that he had the record at his house.

“It deeply moved me and that’s why I started my journey with the talkbox,” Mr Talkbox concluded. “The talkbox was not easy to find and you had to know someone to get that sound.”

But, if you want to make life a lot easier, you could explore one of the talkbox plugins that are available for purchase, which will offer a few choice sound-bites to sprinkle over your track.

However, if you want to try and work with your own unique and malleable talkbox, you'll need to bring in a vocoder plugin.

When choosing a vocoder plugin, most now offer an internal synth, meaning that you do not have to use a secondary synth plugin or hardware to get the vocoder to operate. The synth signal is described as the carrier, with an inputted vocal being the modulatory signal.

For more contemporary talkbox-style sounds, it is also better to avoid plugins with a vintage flavour, in favour of something more versatile. The vintage offerings will probably sound great, but just feel too retro for modern RnB and pop production styles.

Step 1 - Setup

In our example, we are using the EVOC-20 Vocoder, which is included with Logic. Many of the parameters will be available across other plugins, so the settings should be transferable.

Having loaded your vocoder plugin onto an instrument track in your DAW, the first thing to do is to set the modulatory signal path. If you have a microphone connected, you can select this as an input and start vocoding immediately.

You will need to sing/speak at the same time as playing notes on your keyboard. Alternatively, you can select an existing piece of audio as your modulation signal, such as a recorded vocal track, but you will also need to play your keyboard in time with the vocal.

Step 2 - Settings and FX

Make sure that your vocoder plugin is set to operate in vocoder mode, rather than just behaving as a standalone synth. There will probably be a menu or setting for this in your plugin.

Many vocoders employ effects, such as chorus or ensemble, which are used to thicken the signal texture. This tends to offer vintage feel and colouration, so ensure that any similar effects are firmly switched off.

Step 3 - Timbral settings

We can now start to change the sound, bearing in mind that what we are trying to do here is recreate the talkbox sound.

Start by increasing the number of frequency bands available within the vocoder section of the plugin. The EVOC-20 offers a generous 20 bands, so we are increasing this to the maximum.

To replicate a talkbox, we are going to stretch and shift the formant frequency band, so that it highlights the upper-mid frequency range.

We can accentuate this further by narrowing the frequency band (adjusting the slider above the band graphic), and applying a degree of resonance. Important to stress that this is vocoder related resonance, and not that included as part of the synth engine filter section.

Step 4 - Envelopes

You may find that your vocoder offers two sets of envelope setting. One will relate to the vocal input, and the other to the synthesizer. In both cases, we want to adjust them so that there is a relatively short attack and release.

As each plugin will react in a slightly different way, it's advisable to adjust the settings while auditioning your sound. If you have an envelope sustain phase available (on the synth-side of the plugin) ensure that this is set to a full sustain.

Step 5 - Mono

The most successful talkbox sounds work in monophonic form, so switch your plugin to mono mode.

It is this method of utilisation that also gave rise to the common usage of hard-tune, where pitch correction software would create trill-like effects, as and when pitch adjustments do not settle. We don't need pitch correction, as our keyboard is effectively acting as an auto-tune!

Step 6 - EQ

As a final step to altering your sound, engage EQ to highlight upper-mid frequencies, and consider lessening lower frequencies with some form of low shelving.

The flavour of your outputted sound will be heavily dictated by your source synth sound, and in keeping with the tradition of the DX100, the Logic EVOC-20 provides an excellent FM wave source (like Yamaha DX Synths), which act as the perfect starting point for creating the talkbox sound - harsh, digital and bright.

It is definitely worth experimenting with the synth section, although many vocoder plugins may not offer enormous control of this element.

One final point, which might sound obvious, is that getting the perfect sound will take practice.

Playing or programming a vocoder is certainly easier than performing with an actual talkbox, but it still requires time to get to grips with it - so don't be afraid to put in the hours to get the best results.

PLUGIN WEEK 2026: Given the endless bombardment of new plugins, plugin packages and seemingly inescapable plugin subscriptions you might think most DAWs had no onboard plugins at all. But the stock plugin offering in most DAWs is surprisingly good and getting better all the time.

So, we thought we’d take a look at a few classic sounds and show you how to make them with stock plugins. Our DAW of choice is Logic Pro, which has a pretty good selection of stock plugins.

That said, the techniques demonstrated are universal and should be achievable with the provided plugins of most DAWs.

1. Cut-off Reverbs

A great effect that you’ll find in everything from the Prodigy to Skrillex and beyond is the chopped or manipulated reverb. Delivering a more aggressive and tailored ending than a traditional gated reverb, you can achieve this effect with any stock reverb. There are two approaches.

The first and more old-school approach is to set up the reverb as an insert and bounce your sound, including reverb, as audio, then edit the ending to taste. Here we want to add the effect to a block sound on the 4th quarter note. We’re using a long reverb (10 sec decay) and we’ve trimmed this right back so it just overlaps the start of the next bar.

Our second option, and a bit more flexible, is to automate the reverb wet mix level. So, we’re still using the reverb as an insert, but simply blending it in as required. This is a quicker option, and means we can continue to tweak the reverb and blend to taste as the mix progresses.

Finally, if you want to get a bit more creative with the reverb, automation allows us to create a more complex cut up effect. Here we’ve added more automation to the wet mix of the reverb, so the reverb tail works in sync with the bass part below.

2. Screaming Dubstep Bass

Although a modern soft synth plugin might deliver a classic screaming dubstep bass, it doesn’t take too long to build this type of sound from scratch and you can do it with stock plugins. All you need to get started is a virtual analogue synth instrument. Here we’re using Logic Pro ES2, with a single sawtooth oscillator and playback mode set to mono.

The grit for the sound comes from a combination of distortion and a resonant high pass filter. Place the filter first adding medium resonance and steep slope (48dB/octave is ideal). For the distortion some kind of clipper works well - here we’re using Bitcrusher. Increase the drive until you get the required dirty sound - we’ve set this pretty high then reduced the clip level.

If you now play some notes and adjust the high pass cutoff upwards you should hear the basic sound. To finesse this sound we can add some chorus and a short reverb. Here we’ve set the chorus mix quite low, meanwhile for the reverb use a chamber style setting with a medium decay (1.0s), adjusting the mix blend quite low.

The final stage is to automate the high-pass filter cutoff. Fast rising cutoff changes on the note attack work really well and you can then position the cutoff to emphasise the frequency. Once you have these automation moves written, go back and tweak any of the other settings to finesse the overall sound.

3. Formant Shifting

In our final example we’re looking at formant shifting. Popular with all sorts of contemporary artists including Billie Eilish, Cashmere Cat and Jacob Collier, formant shifting changes the resonant frequencies and hence timbre of a sound and is particularly effective for changing the character of vocals - making them sound older or younger - without changing the fundamental frequency.

There is undoubtedly some excellent commercially available software for formant shifting, with Celemony Melodyne at the top of the tree. Even so, formant shifting is available in most DAWs in some form, you just have to know where to look. In Logic Pro, Vocal Transformer is a real time plugin that is great for quick and more extreme changes. For more surgical changes Flex Pitch is the tool, and you’ll find formant editing in the flex options of many other DAWs.

So, how can we use this on a track? There are two popular techniques. First up, we can use it for distinctive harmonies like Billie Eilish’s Bury A Friend. Here we want to process the whole harmony track or tracks, shifting the formants either up or down slightly (200 to 300 cents is usually enough). This works very well if the lead vocalist has also sung the harmonies as it creates more timbral variation.

A more focused option is to use formant changes on specific words or phrases of the lead vocal. This works well on contemporary pop genres (Cashmere Cat is a good example), and is much easier if you have a Flex Pitch style editor. Here we’ve left most of the vocal as it is, and on a few selected words adjusted just the formant amount.

PLUGIN WEEK 2026: Jon Hopkins is a highly influential figure in the realm of electronic music, with a canon of progressive and innovative work that explores the wildest frontiers of techno, electronica, ambient and the spaces between.

Having been championed by Brian Eno (and co-authoring the exquisite Small Craft on a Milk Sea album with both Brian and Leo Abrahams), his career catapulted, working with the likes of King Creosote and Coldplay.

The diverse and often abrasive sounds of his acclaimed fourth album Immunity brought Hopkins to an even wider audience. Its gear-shifts and often invigorating arrangements proved that those who’d rushed to label him as merely a ‘chillout’ act would have to rethink their pigeon-holing;

“I wanted to make things that are emotionally powerful in some way. I think it doesn’t matter what genre it’s in,” Hopkins told The Line of Best Fit.

But Hopkins’ experimental work started long before the public caught up with him, particularly across the gem-filled second and third albums Contact Note and Insides.

Immunity arguably adopted a harder sonic palette, at least in parts. The throbbing distorted bass sounds stood shoulder to shoulder with the beautiful close-miked sound of his own creaky Yamaha upright piano, most often played with the practice pedal engaged for that distinctive felt tone.

After making the record, Jon spoke to the NME:

“It was one of the most amazing experiences I’ve had writing anything and friends who saw me in the middle of it said that I looked like I was in love,” said Hopkins, “I was just euphoric. The only problem is that it’s just impossible to sleep. And the reality is that you’re on your own a lot and I’m a sociable person. What’s brilliant is that once you do finish you can just go round the world playing which makes it all worthwhile really.”

During this Immunity-creation period, he cited a select few working methodologies, alongside some favourite equipment.

Jon would often export audio files in and out of his DAW (which at that time was Logic Pro), manipulating in other packages, before returning them back to Logic. His vintage Korg MS-20 synthesizer often played a central role too, as did the process of complex effects chains; the topic of our tutorial.

A thread that runs through Immunity - at least for many of the harder-edged tracks - is the use of distortion. Jon has always been a big user of Soundtoys plugins, and Decapitator in particular was a go-to as a distortion effect that could really burn up his audio.

For Immunity, this light and shade was pivotal:

“I think you need the abrasive first half, if you were to just listen to the quiet tracks I don’t think it would do it.” Hopkins said to the NME on the album’s often sudden pivots. “I think you need to shake someone out of the state that they’re in in quite an aggressive way, then soothe it."

However, Jon was also a big user of the Bitcrusher plugin, which forms part of the Logic Pro suite. In the audio equivalent of doing exactly-what-it-says-on-the-tin, this plugin degrades the bit depth of the signal, creating an almost gated and distorted texture as a sound decays.

This prevention of a smooth decay, is coupled with a host of audio artefacts which are heavy on the distortion, and can be extended further, with a reduction of sample rate too.

One of Jon’s common tricks was to apply the Bitcrusher effect to a lavishly long reverb tail. His beautiful felt piano - with a resplendently long reverb - is subtly destroyed by the backend use of bit-crushed distortion

We can generate a similar facsimile to the Hopkins sound, while also exploring the medium further. The starting point is to ensure that you have two basic forms of plugin.

Firstly, we need a bitcrusher; many DAWs ship with something appropriate, most likely lurking within the distortion plugin category.

If your package does not have anything appropriate, TAL Software offers a freebie, simply called TAL-Bitcrusher.

Alternatively, Decimort 2 from D16 Group, is one of the most highly regarded paid options, for just €59.

Secondly, we are going to need a good reverb, and one that is capable of generating long reverberation times.

Eventide’s Blackhole plugin has always been the gold standard in this regard, however the ever popular Valhalla Super Massive (or indeed, many other Valhalla ‘verbs) will also do an excellent job.

If you want something more versatile, consider Lunacy Audio’s excellent Space; it is convolution based, and great for other reverberation tasks, but is also capable of generating some very long reverb tails.

Step 1. Creating Your Track

As a starting point, we are going to keep things traditionally ‘Jon’, by recording a track with a felt piano sample, and some very simple chords.

There are a multitude of different felt piano samples available on the market, which include the superb Soniccouture Deep Felt, sampled from a Steinway Essex upright instrument. However, one of the longest standing free examples comes from Spitfire Audio.

Included as part of its freebie Labs series, Soft Piano has been used on more cinematic and TV scores than we care to think of!

It’s basic, but exceptionally good for a freebie.

For the best effect, try working at a lower tempo, and give your chords time to play out and settle.

Step 2. Adding Reverb

It's time to add our first effect, although you can approach this in two different ways.

Traditionally, you can simply add an effect to your channel strip, however it can be very useful to have control over the amount of signal being sent to the effects chain, so applying some form of auxiliary or bus will offer maximum flexibility.

We are working in Logic, and will assign a bus send to an empty auxiliary channel strip. Be sure to turn up the Send amount, and load in your chosen reverb on the related bus channel. We are using Eventide’s Blackhole reverb in our example.

As our reverb is on an FX bus, adjust the Wet/Dry mix, so that the signal is fully effected (or Wet).

Step 3. Adding Bitcrusher

Now it's time to add the Bitcrusher effect. We are keeping it basic, by using Logic Pro's included Bitcrusher, but many of the parameters will be the same across other plugins.

The primary control that we want to explore is the Resolution control, which allows us to alter the bit-rate depth.

Lower values will result in a harsher form of distortion, with higher values being more pleasant to listen to.

You will also notice that lower values take effect far quicker than higher values, so you may choose to alter this value, dependent upon the space between each chord that you have played in your track. Adjusting the length of the reverb time will equally change the Bitcrusher’s effect on the signal.

You may also choose to adjust the level of the Aux/FX chain track, as it may be overpowering against the original piano track.

As always, make sure you listen to the track with objective ears, and adjust your levels accordingly.

Step 4. Additional Elements

Just like Hopkins himself, it is always a great idea to draw inspiration from music convention, but then follow your creative instincts into your own direction, making it unique to you.

With this in mind, there are a couple of details that are worth exploring.

Firstly, in slower tracks, you can extend the reverberation through the use of delays. You could try inserting a delay, with a healthy amount of feedback, immediately after your reverb in the effects chain.

You can also try adding a second reverb plugin, which allows further reverberation of the bit-crushed audio. This would go at the very end of the effects chain and, unlike our previous setting, would need to be adjusted so that you can hear the dry signal, as well as the wet.

Of course, you can tryout multiple instances of all of these plugins to achieve your own unique style and effect.

Always keep in mind though, that this process will be more successful on slow moving parts, or with intermittent and occasional percussive notes.

PLUGIN WEEEK 2026: Heartthrob, crooner and fashion renegade - is there anything Harry Styles can’t do? Apparently not. As demonstrated with his recent hit, Aperture, Styles has become a keen embracer of the sonics of minimal electronic music.

It’s a track that is moody and atmospheric, yet driving and bouncy, all while leaving enough room for Harry’s vocals to shine.

Eschewing traditional instrumentation, Styles' Aperture has more in common texturally with something like Nathan Fake - The Sky Was Pink (James Holden Remix), combining sculpted white noise, atmospheric chords, lofi synthesised beats and heavyweight bass.

One might also compare it to the experimental electronic work of performance artist Laurie Anderson with its processed vocals and repeated voice nuggets.

It’s certainly a far cry from Harry’s more traditional work as part of pop icons One Direction, yet crucially, its electronic elements sit tastefully in the mix with Styles’ vocal remaining at the center.

The combination of Harry’s familiar voice and unusual electronic backing saw the song become a huge hit with fans, who no doubt appreciate the track’s unorthodox production even if they’re not fans of the minimal classics that seemingly inspired it

If you’d like to join Harry in this minimal madness you’re in luck - the effects heard in Aperture can be achieved easily with even straightforward soft synths. So let’s break down the track to find out how it weaves its synthetic magic.

1. Pumping organ chords

The foundation of Aperture is its synth-based, organ-like chord progression, which ducks in volume on each beat much like Eric Prydz’ Call On Me.

To create something similar, we can begin with the Arturia Jup-8 V’s Combo Organ preset. Here’s how it sounds playing a I - V - ii chord progression.

This organ preset doesn’t have the smeared, dreamy quality of the Aperture synth, so let’s smooth it out a bit by changing VCO-2 to a sawtooth, dropping it down an octave, reducing the LFO speed and VCO modulation amount, and also turning down the Envelope 2 to filter cutoff.

This gives us a more atmospheric, wistful sound that’s closer to Aperture.

Next we need to add that sidechain-style pump. We can be sure that this isn’t actually sidechain compression, because the ducking doesn’t match the non-4x4 pattern of the kick drum, so let’s use a volume modulating plugin (in this case Ableton Live’s Auto Pan) to duck the volume on quarter notes.

With the effect applied, we get that driving ducking effect that gives the song part of its propulsive energy.

2. Techno kicks

Aperture’s kicks have a techno feel. However, they don’t follow techno’s standard 4x4 rhythm, instead opting for a pattern reminiscent of another early 21st century dance anthem, M.V.P. - Roc Ya Body (Mic Check 1 2).

Creating a kick pattern like this is straightforward: starting with a Roland TR-909 kick drum sample, we sequence it on the first three beats of the bar, then add kicks on the 12th and fifteenth 1/16th notes.

Because the 16th notes don’t perfectly align with the pure 4x4 of the sidechain-esque pump, it creates an interesting rhythmic interplay with the ducking synth.

3. Evolving plucks

Another element that helps drive the track are rapid 16th-note synth plucks. These have a noise element a little like Kelis’ Acapella, with filter modulation opening and closing the sound to create a sense of development.

We can make a version of this sound by starting out with a PWM-style envelope with multiple unison voices, and a low amount of Detune. A noise oscillator provides the aforementioned noise element.

This sounds kind of messy as-is, but we can make it tighter and more energetic by turning down the decay and sustain on the amplitude envelope. We’re going for a really tight decay time here, which helps give the part a propulsive feel.

The final touch is a band-pass filter that’s modulated by two free-running LFOs, one with a period of a quarter note, the other with a period of 8 bars.

Again, this ducks on quarter notes, so let’s add another Auto Pan effect to get that pumping feel.

4. Bouncy Bass

Big, round and bouncy, a little reminiscent of San Soda’s Juno Love, Aperture’s bassline is a great example of 'less is more' in action. In fact, it’s just the tonic note playing on one octave, then the octave above. Simple, yet extremely effective!

The sound design of this bass is simple too. All you need is a square oscillator with a low-pass filter closing over it with a medium length decay time, and just a little portamento glide.

5. Lo-fi percussion

Taking influence from minimal techno masterpieces like Richie Hawtin AKA Plastiman’s Spastik, Aperture continues to make use of white noise in its percussion department.

Making the lowest of lo-fi hi-hats is a breeze: simply use a single white noise oscillator, and turn down the amplitude envelope’s sustain and decay so that you just get a little burst of noise on each note.

Now, let’s hear how all these elements sound together.

So there you have it, five production techniques employed by Harry Styles when he took the sounds of minimal techno pop-ward.

If you’d like to hear more of the original minimal sound, check out The Sound of Minimal Techno on Spotify.

There's a lot you need to learn if you want to make it as a remixer, but these half dozen bite-sized tips are a great place to start.

Remixing is all about extracting the parts and elements from a track and then rearranging them – and adding new parts of your own – to form a new mix.

Let's get started.

1. Beyond pitch and time

Timestretching, pitchshifting and contorting audio and MIDI tracks to fit with chords are essential skills for remixing and bootlegging. Take the time to learn how your DAW handles these tasks and investigate the many plugins out there designed to make everyday remix chores even easier.

2. Layer by layer

If a track’s made up of layers, with one new element added every few bars, there’s a chance you’ll be able to duplicate the channel and use phase inversion to isolate elements from each other.

That's to say that you can take a sample of audio from an earlier part of the track then phase invert it alongside the next part of the track. The two 'original' samples will then magically mute each other out, leaving only the audio of the new addition.

This is a great technique for extracting vocals or new synth parts ready for further use or manipulation elsewhere in your remix.

3. Straight down the middle

With a mid/side plugin like Voxengo MSED, you can mute the mid or side signals, helping isolate certain elements of the mix.

The mid information comprises any audio that’s exactly the same in both speakers at any given point (mono), whereas the side signal represents what’s different in both speakers. The point being, as you’ll see, certain mono elements can be extracted from stems or full tracks.

4. Sound conversion

Many DAWs have some form of Audio to MIDI conversion, meaning you can take an audio file and spit out a MIDI representation of it. Naturally, this is great for remixing, helping you quickly layer or replace recorded parts with virtual instruments.

Want to replace one sound with another? This is how you do it: Turn the audio into MIDI, then use that MIDI track to drive your own synth sound or sample.

5. Beats working

Getting a great drum sound is an essential part of making a great remix.

Listen through your work-in-progress mix to find individual hits you can chop out and lift. Load them into a sampler or arrange them on a track and create your own rhythm additional rhythm tracks with all the polished pro-quality sound of the original source.

You can also isolate individual hits from a full beat using phase-inversion, mid/side processing and manual or automatic slicing. Again, look into your DAW’s particular tools in this department, as each has its own ways of doing things.

6. Groove is in the art

Groove templates let you steal the exact feel (ie, the detailed timing and dynamics) of an original piece of audio. By imprinting the exact swing or ‘groove’ of the track you're remixing onto your own audio and MIDI clips, you can make your re-imagining feel a little more authentic.

It's therefore possible to extract the feel and timing of one track and totally superimpose it onto another. Or give your lifeless work-in-progress track a whole new lift by 'borrowing' the timing and feel of someone else's success.

Don't fear the tab. Tab is here to make reading music simple and, while at first glance it may SEEM just as confusing and fiddly as conventional music scoring, it's actually a WAY simpler way to explain which notes to play (and where to put your fingers).

Plus there's a whole bunch of picking and bending instruction - essential for modern playing - that old-fashioned music notation simply doesn't provide.

Tab is short for tablature, and it basically shows where notes should be played on your guitar's fretboard. i.e. Rather than being a list of notes you need to play (and letting you figure out the realities of how to do that) it's literally a map of where to put your fingers.

Tab often appears underneath conventional music notation as six horizontal lines that represent the strings of the guitar, from the sixth (thick) string at the bottom to the first (thin) string at the top.

It's like this so that if you sit with your guitar in your lap and look down at it, the tab on the page (or on the screen) will match the strings on your guitar neck.

Thick low E closest to you (at the bottom of the tap)… Top E furthest away (at the top of the tab). Simple.

On these lines, numbers represent the frets where you should place your fingers. As per our example above, a D note on the 2nd string, 3rd fret will be shown as a number ‘3’ on the second line down on the tab.

Unfretted strings (or 'open' with no fingers on the fretboard) are shown with a ‘0’. i.e. No fret needed - just play the string.

The key and time signatures are shown in the traditional notation. A helpful timestamp may also tell you where in the actual audio track this bit of music appears. And a tempo marking is expressed in beats per minute.

Fret boxes: chords, scales and capo notation

Hand labelling

Of course, with tab telling you where to put each finger on the fretboard, you're going to have to know which finger it's talking about…

Here are the tab abbreviations. On your fretting hand that's 1, 2, 3, 4 and T for thumb

And on your picking hand that's p (thumb), i (index), m (middle), a (annular), c (little finger)

Chords mapped on the fretboard

This is a fretbox.

The fretbox diagram represents the guitar’s fretboard exactly, as seen in the photo. Now, by placing numbered 'fingers' on the fretbox it can be used to quickly show you where to put your fingers in order to play chords or a scale.

Playing a chord

This diagram shows you how to play a G chord. The ‘o’s are open strings, and a circled number is where to place your fretting hand finger. A black ‘o’ or circled number denotes the root note of the chord (here, G).

Simply place the right fingers where it says… And strum… Done!

Using a capo

If you see a blue line on the diagram that represents where to place a capo.

In our example above, for this A chord, the capo is placed at the 2nd fret. Note that capos – by effectively shortening the neck of the guitar – change the fret number ordering.

A capo on the 2nd fret moves everything two frets up. So here, the original 5th fret now becomes the 3rd fret. The 7th fret now 5th fret, and so on.

Basically the capo can be considered as being the 'nut' at the end of the neck and so all counting starts from there.

Capo notation

Here, the chord looks like a C in the tab, but the capo on the 2nd fret raises the pitch to make it a D. The 2nd fret capo’d notes are shown with a ‘0’ in the tab as if they were open strings.

Tab showing a scale

Here the fretbox diagram illustrates the fret hand fingering for the A major scale.

It's using black dots for root notes and red dots for other scale tones. Meanwhile the photo shows part of the scale being played on the fourth string with the first, third and fourth fingers.

Guitar techniques: Picking

Down and up-picking

Tab does its absolute best to tell you as much information as possible – even down to how to pick notes.

Here symbols under the tab tell you the first note is to be down-picked and the second note is to be up-picked.

(Remember that '7' in our example is the fret number you need to hold down rather than being a seventh finger…)

Tremolo picking

There's even symbols for tremolo playing. Here each of the four notes are to be alternate-picked (down- and up-picked) very rapidly and continuously.

Palm muting

And here's the symbol for palm-musting notes. A palm-mute is acheived by resting the edge of your picking hand palm on the strings near the bridge saddles which stops them ringing so brightly and gives that distinctive 'muted' sound.

Pick rake

Here you've got to drag the pick across the strings shown with a single sweep. This is often used to augment a rake’s last note.

Arpeggiated chord

And here you have to play the notes of the chord by strumming across the relevant strings in the direction of the arrow head.

Fretting Hand

Hammer-on & pull-off

Let's get into some more techniques that tab can features.

For the hammer-on and pull-off here you must pick the first note then hammer down on the string for the second note. Pick the third note and pull-off for the fourth note.

Note trills

And for a trill, after picking the first note, rapidly alternate between the two notes shown in brackets using hammer-ons and pull-offs.

Slides (glissando)

A glissando (or slide) looks like this.

Pick the first note and then slide to the next. For the last two notes pick the first, slide to the next and then re-pick it (that's the 'RP' symbol).

Fret-hand tapping

Sound the notes marked with a square by hammering-on/tapping with your fret-hand fingers, instead of picking.

Fret-hand muting

Here, the X markings represent notes and strings that are muted by your fret hand when struck by your picking hand.

Bending and vibrato

Bend and release

Fret the first note (here, the 5th fret) and bend up to the pitch of the bracketed note, before releasing again.

Re-picked bend

Bend up to the pitch shown in the brackets, then re-pick the note while holding the bent note at the pitch shown.

Pre-bend

Silently bend the string up from the 5th fret (PB5) to the pitch of the 7th fret note, pick it and release to the 5th fret note.

Quarter-tone bend

Tab really has got everything… Here you have to pick the note then bend up a quarter-tone (a very small amount). This is sometimes referred to as a ‘blues curl’.

Vibrato

Vibrato is an essential playing technique and tab has got you covered.

Here it's telling you that your fretting hand should vibrate the string by small bend-ups and releases. You can then exaggerate this effect to create a ‘wide’ vibrato.

Harmonics

Natural harmonics

Pick the note while lightly touching the string directly over the fret indicated. Get your technique just right and you'll get a lovely chiming harmonic.

Artificial harmonics

Fret the note as shown, then lightly place your index finger directly over ‘x’ fret (AH ‘x’) and pick (with a pick, p or a).

Pinched harmonics

After fretting the note in the triangle, dig into the string with the side of your thumb as you sound it with the pick.

Tapped harmonics

Place your finger on the note as shown, but sound it with a quick pick hand tap at the fret shown (TH17) for a harmonic.

Touched harmonics

A previously sounded note is touched above the fret marked TCH (eg, TCH 9) for it to sound a harmonic.

Vibrato bar/whammy bar

Whammy bar bends

Here, the note is picked as shown, then the vibrato bar is raised and lowered to the pitches shown in brackets.

Scoop and doop

And you can even scoop and doop…

Scoop: depress the bar just before striking the note and release. Doop: lower the bar slightly after picking note.

Sustained note and divebomb

A note is sustained then the vibrato bar is depressed to slack. The square bracket indicates a further articulation.

Gargle

Sound the note and ‘flick’ the vibrato bar with your picking hand so it ‘quivers’. This results in a ‘gargling’ sound!

Whammy bar vibrato

Gently rock the whammy bar to repeatedly bend the pitch up and down. This sounds similar to fret hand vibrato.

Others

Pick scrape

The edge of the pick is dragged either down or up along the lower strings to produce a scraped sound.

Violining

Turn the volume control down, and play the note(s) with the volume off. This removes the initial 'pluck' from the sound. Then turn the volume up for a smooth 'pluck-less' fade in.

Finger numbering

The numbers in the traditional notation refer to the fingers required to play each note.

pima directions

Any kind of fingerpicking requirements are shown at the bottom of the tab notation.

Pick-hand tapping

Tap (hammer-on) with a finger of your picking hand onto the fret marked with a circle. Usually with ‘i’ or ‘m’.

And that's it!

Don't stress over some of those more advanced techniques. We've included them here for completeness and – chances are – you'll come across some of them only very rarely.

Spend time nailing the basics and practicing the tab on songs you're familair with.

Then, as your confidence grows you can tackle more complex tab and always come back to this guide to remind yourself of what some of those little squiggles really mean.

Want to learn more. With tab under your belt why not try learning a few open-position guitar chords?

The pure-pop appeal of Olivia Rodrigo’s euphoric Drop Dead is undeniable, even if the lyrical approach feels incredibly personal, laced with all the idiosyncrasies of the current social media age.

I’ve been playing this track a lot, and have become fascinated by its colourful, evolving arrangement. There’s so many elements of this Dan Nigro-produced banger that are crying out for deeper exploration. The opening, more electronic, section of the song has particularly captured my attention.

This section provides us with a beautiful soundscape and backdrop for the vocal, which is continually punctuated by a synth bass which is deliciously trance-like in tone.

The upper elements rely upon reverse and delay effects, applied to an excessively high-pass-filtered synth pad, but it is the indulgent bass part which I really love.

It's such a retro - yet modern feeling - sound that I can imagine using it in a range of different musical contexts. But first, we need to remove it from Drop Dead…

Up for a challenge, I thought it best to to create this bass sound from scratch, element-by-element until we get something fairly close to the original.

However, with a little re-working and re-voicing, the same sound can be exploited for tracks reminiscent of those trance-floor classics, such as Underworld’s Born Slippy, as well as your own dynamic pop hits!

Firstly, we're going to need the right tool for the job - namely a synth…

1. Choosing a synth

In choosing the best synth for the task, we have to form an opinion from the sound itself.

There is a sense of close tuning - almost phasing - in the sound. This could be replicated with specific plugin effects, but going straight to the source of the sound indicates that the ideal partner is a synthesizer plugin which offers three oscillators.

There is less choice of synth in this domain, as a majority of synths offer only two oscillators, but it does suggest the use of a Minimoog-inspired sound source.

Options such as Logic Pro's ES2 are viable, but with so many Minimoog emulation plugins available, something with the classic Moog architecture is your best solution as a first call, if you have one available.

I'm going to be using one of the most cost-effective options; Cherry Audio’s superb Miniverse plugin.

2. Start from a clean slate

Whichever model of plugin synth you use, it makes sense to begin with a default patch or setting.

Most plugins have some form of ‘new’ patch option, and in the case of Miniverse, you simply need to click the New button, which is located top-left of the plugin window.

3. The oscillators

I start building the patch with the use of all three oscillators.

We need to set all 3 to a range of 16’, setting the first two oscillators to a regular sawtooth. The third oscillator can be set to either the reverse sawtooth, or you can generate a slightly different colour/timbre by experimenting with one of the pulse waves. They all provide a slightly different tone, but will pleasantly alter the texture as a consequence.

One thing to always be aware of is that some Minimoog plugins adopt the traditional ‘Osc 3’ button, which means that the 3rd oscillator does not track the pitch of the keyboard, unless the setting is changed.

Make sure this is deactivated, as you definitely want Osc 3 to respond to your playing.

4. Some light de-tuning

Sticking with the oscillator section, we need to make some subtle alterations to the tuning of both Oscillator 2 and Oscillator 3.

As we are working within the plugin and digital domain, we need to engineer a small degree of de-tuning, so that the oscillators do not phase.

There is small degree of phasing taking place on the track, which adds to the overall sonic character, but making some subtle adjustments will hone this to perfection.

The tuning of Oscillator 1 is fixed globally, so move to the tuning pot of Oscillator 2, adjusting it by a minuscule amount, to a value of +0.02. By similar token, adjust Oscillator 3 to a negative amount of -0.02. This should create a thick and rich texture.

5. Mix balance

Moving onto the next section, make sure that all three oscillators are activated in the Mixer section, and that they are adjusted to the same volume.

We have set all three oscillators to a volume of 8 out of 10 (80%), but you can obviously adjust this to suit whatever you're working with.

6. Filter settings

This bass sound benefits from a relatively heavily filtered setting, with accentuation provided by the filter’s envelope.

Moving to the Filter section, we have set our low pass Cutoff pot to a value of 30%, ensuring that the Filer Emphasis (resonance) value is set to 0.

7. Setting the Envelope

Turning our attention to the Filter Envelope, we want a really harsh attack time, which we have set to 0, followed by a decay time of around 400-450 milliseconds, which is fairly snappy. We don't need any sustain, so we have set this to a value of 0.

In order for the filter envelope to take effect, we need to ensure that the Amount of Contour pot (A.K.A Filter Envelope amount) is set to an appropriate value. Try something in the region of 40% as a starting point and adjust to suit your track.

The amplitude envelope (Loudness Contour) requires very similar settings, although we have chosen to set the decay time to a value which is closer to 600 milliseconds. Similarly, we have set the sustained value to 0.

One final point to check, with regard to Minimoog-style instruments and the envelope section, is that the Decay button is switched on, which means that the envelopes will provide a release phase, and not a heavy release gate. It’s often located bottom-left on Minimoog plugin windows.

8. Distortion and delay

The synth sound itself is now ready to use, but there are a couple of additional elements which are fairly vital, if you want to create a similar texture.

Firstly, Minimoogs always respond well to a degree of saturation or distortion, so we have applied a very small amount of Soundtoys Decapitator to our sound, which will introduce a lovely grittiness and bite, at the front of each note.

Even more important though, is the selection of an appropriate delay. Of course, you can use any delay you like, but the example used here (and on the track) is relatively clean.

This means that the graininess of a tape delay style plugin, might not be what you need. We have used a basic delay (Echo) from Logic, which is designed to keep the signal clean and unaffected, apart from the obvious repetitious delay element.

The delay needs a fairly generous Feedback amount, while the Wet volume control is set very low. This means that the delay is audible, but does not get in the way of the signal. We have also sync’d the delay with our DAW, ensuring that we have 1/8th note delay repetitions.

Repurposing

With a little further imagination, and a switch to a polyphonic synth setting, this Drop Dead-inspired patch is perfect for punctuating chords which repeat using delay.

Use it much like a pad sound, and you can always soften the envelope attack, if it feels overpowering or heavy as you play each chord.

SYNTH WEEK 2026: The wonderful thing about the emergence of synthesizer technology in the '70s and early '80s is that it gave rise to many fantastically engaging stories, which have since entered the folklore of the synthesist’s almanac.

One of the most often recounted moments could be described as ground zero for Gary Numan.

So the story goes, he walked into a studio with his band, Tubeway Army, with the intention of making a punk album. Unfortunately for punk, there was a Minimoog sitting in the corner of the studio, set up with a patch that literally made the room shake - and the rest (as they say) is history!

In 1979, Numan released The Pleasure Principle, the first album under his own name and away from his band. While there were clearly band-driven elements scattered throughout the album, its reliance on synthesizers was groundbreaking in pop culture - with no track exemplifying this more clearly than its lead single, the (UK) chart-topping Cars.

Surprisingly, according to Numan, the iconic riff of Cars was actually first written on bass guitar.

“I just wanted to do pop music, and thought Cars was here today, gone tomorrow,” Numan told Chaos Control. “I don’t know what happened with that, the fact that it’s done so well over the years - I never quite figured it out, really. It was the quickest song I ever wrote. I wrote it on bass guitar, I didn’t even write it on the keyboard, and it took about 10 minutes, the tune and the lyrics. The keyboard part came later, I mean the main melody and arrangement of the song and lyrics. It was about 10 minutes, beginning to end.

It’s the distinctive sound of that iconic main riff that we’re digging into today, which also forms part of the melody for Numan’s vocal.

When we focus on this element of the song, it's actually quite difficult to hear exactly what's going on because, despite the overall simplicity, it's tricky to filter out the additional elements, such as the bass and drums, but we can certainly derive some useful detail.

First of all, the main riff has been recorded twice, once for the left channel and once for the right. The two sounds used for this riff are both generated by the Minimoog (a Polymoog with the ‘Vox Humana’ patch, was deployed to create the additional swirling lead sounds which enter the mix later).

The twin-Minimoog-driven main riff provides a solid musical foundation, whilst also providing a feeling of movement in the sound. Both sounds are quite ‘reedy’ in harmonic makeup, suggesting that they are predominantly formed from a pulse wave.

The Cars riff patch is a sound that could be recreated using any synth that offers subtractive synthesis, armed with two or three oscillators. For this exploration, we are going to create a soundalike patch using Cherry Audio’s cost-effective Miniverse plugin, which is a hugely exacting software emulation.

One of the unique features of the original Minimoog is that it did not offer pulse width modulation (PWM).

There were post-market customisations available to add this, but Moog’s answer to the issue was to provide three different pulse wave shapes, within each oscillator section. It is these wave shapes that are fully exploited for the sound of Cars, with a slight variance between the left and right channel.

If you would like to re-create the sound in your own DAW, then you must start by creating the patch for the left channel. It's important to note that you should load your plugin synth in a mono instance, rather than stereo, as this will allow you to pan the synth fully to your left channel.

How to replicate the Cars main riff sound

Using Oscillators 1 and 2, set these to a middle frequency range, such as 16’.

As a starting point, select the first and second pulse waves respectively, making sure that the volumes are set roughly the same. If your chosen synth offers PWM, alter the pulse width settings so that the two oscillators differ slightly in tone.

Depending on the synth you are using, it will be worthwhile creating a small amount of de-tuning. In the case of a Minimoog-in-software, this will mean changing the tuning of Oscillator 2, as the tuning of Oscillator 1 is fixed.

The other related elements to the sound are fairly simplistic, but will once again be open to variation, according to your chosen synth. The filter Cutoff Frequency should be set very close to fully open. In our experience, software versions of the Minimoog often include a cutoff frequency that opens wider than the original hardware, so you may need to reduce the brightness by a small degree. There is no resonance (aka emphasis) applied to this sound.

The filter envelope doesn't play a part here either, but the Loudness Contour (aka Amplitude Envelope) certainly does.

The patch fully sustains, and adopts a very fast attack time, and a rapid decay/release time. If you haven't done so already, switch your synth to monophonic mode or single voice, which will also limit any release phase time to the end of phrases.

That’s pretty much it for the left channel, but the patch for the right channel adopts very similar sensibilities, in fact, it would be worthwhile copying the L/H patch over to the R/H channel, for ease and speed of setup.

The subtle variation between patches takes place within the oscillator section again, where you need to select different pulse waves, providing a sort of ‘ying-yang’ effect between the two channels.

There is also a thought that a triangle wave is used in the R/H patch, which is something worth trying and applying. If you have a third oscillator available, use this and subtly apply a triangle to the existing sound. Much like our first patch, you will probably need to apply a little detuning, to avoid unwanted beating, caused by precise pitch accuracy.

All the other settings, relating to filter and envelope remain the same, although you could subtly tweak these, to create a slightly different tone and envelope tail.

If you haven’t done so already, pan the first patch hard-left, and the second patch hard-right. This will create a gloriously thick and complex texture.

Bussed for Phaser

Having created the two patches, you can record the riff on one part, and copy it to the other part. However, if you would like to adopt the purist’s approach, play and record the riff for each channel, without using quantize. The subtle timing discrepancies between parts will add to the overall flavour, getting closer to the original.

Like many synthesizers from this era, Minimoogs were very reliant upon external effects, and no other effect was more exploited on this album than a phaser.

As most DAW’s offer phasers in one guise or another, bus both of your synth patches to a single bus channel, and apply phaser to the bus. This is certainly an area for experimentation, as you could try various forms of stereo phaser, with variance of setting to get the perfect effect.

Other Elements

While Gary Numan’s music from this era was intrinsically linked to synths, he still made considerable use of conventional instrumentation.

The drums are provided by an acoustic drum kit, and the bass is provided by a bass guitar, however there would be no issue with using a further Minimoog-style patch, to provide a bass one octave below the riff we have been discussing here.

In fact, in the spirit of experimentation, you could use a third oscillator tuned an octave lower, to provide a thicker texture, all sounding simultaneously.

“I think to this day that Cars remains the most poppy song I’ve ever written” Numan reflected on his unlikely hit in an interview with Billboard, “I do not think I have a talent for pop music. It’s too light for me, for my taste in music. It’s too…happy. I’m just not very good at it! I’m much better if I slow it down a bit, make it a bit menacing, add a hint of mystery or oppression, I’m in my element at that point!”

SYNTH WEEK 2026: It's likely that we're all guilty of periodically shunning our existing repository of plugins and VIs, and splashing out on a shiny new synth, hoping it’ll provide us with a fresh, inspiring new sound for our latest project.

Meanwhile, back in the land of reality, we all know that there's a lot of hard work and graft that goes into creating the right synthetic sound for a track, let alone constructing a good mix.

Often it's best to stick to what you know - and focus on widening your knowledge.

Extending your synthesizer plugin techniques doesn't have to be demanding, but it might require a slight change in your working practice.

So, if you fancy taking your synth sounds to the next dimension, then follow this guide to making the synths in your collection go further - and not being slowed by option paralysis!

1. Tweak a preset

There’s not a producer on the planet that doesn’t load up a synth for the first time and then trawl through the preset patches, but how often do you find something that immediately fits the bill? Maybe you find something cool, then re-shape your track to suit the sound you’ve found. We’re here to tell you that it doesn’t have to be this way!

Don’t fear the preset, but also don’t be afraid of tweaking the basic settings, such as the filter or envelope. You might also find that switching up other elements within the patch drives your work in a better direction

2. Start a new patch

Bored of your usual preset options? Most synths offer a perfectly-placed button on their interface, inviting you to craft a new patch of your own. The benefits and rewards to be found here are huge, as this process will allow you to create a bespoke sound to suit your very own requirements.

Designing a sound from the ground up, is one of the most fulfilling aspects of production, and not something to be feared.

3. Turn off the effects

Find us a plugin synthesizer, and we will find you an included set of effects. Locating the effects page will reveal everything from chorus and delay, to the ubiquitous reverb, and probably more besides. However, these onboard effects may not be as good as the third-party effects, which are specifically designed for purpose, residing within your DAW.

Sure, many of the effects may add something to the preset sound, but don't be afraid to switch these off, and employ your own selections. It’ll also provide greater flexibility in your working.

4. Multi-layer parts in mono

More complex synths often allow multi-timbral usage, meaning that you can load a single instance of a synth, but use it for anything up to 16 different sounds, at one time.

This puts less loading on your computer. However, you don't have to settle for a single channel of sound, thanks to the ability to set up multiple outputs.

Organise these to apply different mix settings for each sound.

Total flexibility in your mix, without the need the load more than one synth plugin.

5. Automate!

How many times have you seen a synth-genius playing live, carefully massaging settings as they play? Extend this philosophy to the ambidextrous world of the DAW, and you can tweak a series of different settings all at the same time, placing them under automated control.

Moreover, this will add a wonderful layer of developing interest to your synth patches, that can make your tunes feel less repetitive.

6. Get yourself templated

Next time you start a project, take a look at a small selection of plugin synths, the ones that you tend to use the most.

If you have a go-to synth for basses, set it up in a template, so that every time you start afresh, you can get cracking with creativity, and not have to continually think about plugging in your usual set of sounds. Not only will it save you time, it will provide a common thread from one track to another, which might be beneficial for an entire project.

7. Arpeggiate

Part of the package that accompanies most plugin synthesisers, is the use of onboard sequencers and arpeggiators. These notes may be preordained, but they don't have to stay that way

If you can utilise the sequencer within the plugin, change the notes and settings to suit the developing rhythm of your track, or alternatively, look at using similarly-equipped plugins residing within your DAW.

8. Modulate

If there is one technique that occupies the very heart of synthesis curiosity, it's modulation.

Some synths can take modulation to extremes, with the ability to modulate every aspect of a synth’s sound, from wave shape to LFO speed and depth.

Experiment fully with the options available within a plugin, and you may find that the odd happy accident drives your creativity in a new direction.

9. It doesn't have to be software…

It doesn't always have to be about plugins! Most DAWs are fully equipped to handle an incoming signal from devices including external synthesizers.

Moreover, they are equipped to handle MIDI, making light work of the ability to patch in a hardware synth.

If this becomes something that you do regularly, pop your hardware device into your template, then it will always be ready to go when creativity takes a hold.

10. Learn, learn, learn!

We get so bombarded by products and deals that we're all guilty of having far too many plugins on our systems, in fact you may have so many - you have forgotten what they all do! (certainly a problem I can vouch for, Ed).

Herein lies one of the biggest problems with production in the current age, but if we take a step back, we could learn a lot from previous synthesis of pioneers.

Next time you load up a synth that truly inspires you, set yourself the goal of really learning how it works - just as previous synth pioneers had to do with their limited set of hardware synths.

Many aspects of software synths may appear to be simple enough - and you may feel you've got a handle on their essentials - but the chances are there are a ton of complexities which remain uncovered, and could provide you with that extra push over the sonic cliff. You could try reverse engineering a preset sound, or better still, create a sound from scratch, and take it in a direction that you wouldn't normally go.

If you find yourself scratching your head, reach for the online manual, and take a bit of time out to scrub up on what a particular parameter does to induce a particular sound. You probably have everything you need for a track within just one synth. Often, it's not even the most expensive.

Scales might sound daunting at first, especially with names like Dorian, Lydian and Mixolydian – but don't be deterred. Follow along as we break down three essential scales into digestible bite-sized chunks that will help you play along to your favorite songs and save you in a jam scenario when your fellow bandmate says a song is in a particular key.

Scales usually revolve around a single chord shape. You then just add extra complementary notes around them to complete the scale – we'll show you how to do this below.

Once you've learnt all of the notes in the scale pattern, there are basically no wrong notes – this is perfect for jam situations, whether improvising over a song or just a few simple chords that you've looped together.

Practising scales for a few minutes every day will also hone your alternate picking technique, help make your fretting hand more accurate and - with the help of one of the best metronomes - improve your timing and stay in the pocket of a groove.

All of this will result in two things: your playing will be more accurate and professional sounding, and you will almost certainly become capable of playing faster.

It's important that you don't play scales just to get faster, though. Speed is a happy by-product of trying to play more accurately. Even if you don't fancy becoming the next über shredder, practising scales is still important.

Make sure your guitar is in standard EADGBE tuning – the scale won't sound quite right if one or more strings are out of tune, then let's jump in to clear up the confusion around those pesky scales and you'll be improvising brilliant solos in no time!

1. A major pentatonic 

Everyone knows the A chord. As you play it here, make sure it’s the cleanest A chord ever heard!

The major pentatonic scale is great for creating bright, melodic lead lines. It also contains the A chord.

The idea here is to see how the chord, scale and lick all relate to each other – the three shapes are pretty similar. Play the A chord with a first-finger barre. Your third finger should fall into place.

Play our first track to see how the scale could sound over the top of some basic chords, then when you're ready load up the second track and use the tab above to follow along using just the D, G and B strings.

Once you've got the hang of it why not experiment with other notes using the A major pentatonic scale diagram above – don't worry, there aren't any wrong notes to play so there's no incorrect way to play them – get creative!

2. A minor pentatonic 

Play each note one by one to check they’re ringing clearly – adjust your thumb position if needed.

Notice how the notes of the chord are found within the scale. It means they’ll sound good together.

Jimi Hendrix used to thicken his lead lines with doublestops. This lick shows how he might have combined them with a bluesy bend. Try branching out your ideas by using other notes from the chord and scale.

Rather than sliding your fretting hand up from the 5th fret to the 7th in this scale you could get creative and instead try bending the note at the 5th fret one whole step instead and this will give it more of a bluesy feel.

Alongside famously playing a right-handed Stratocaster upside down to suit his left-handed playing, Jimi was also know for playing bar chords with his thumb over the top of the neck to fret the low E string – experiment with this thumb over style and fretting the bar chord with your index finger to find which you find more comfortable.

3. Mixolydian 

Notice how the dominant 7th chord sounds unresolved, like it needs to move to another, more final chord.

Look closely! Once again the notes of the chord are found in the scale. Jam around the two together.

This laid-back melody is inspired by the North Mississippi Allstars’ softer moments. Once again, play close attention to the lick’s feel and try some more ideas with other scale and chord notes.

Learn more about the minor pentatonic

Take it further: Finesse your timing with a metronome

A metronome is a device used to give a steady, constant pulse for you to practise to. It is also referred to as a click track, especially in studios. Metronomes are fairly cheap (they're available for as little as £15), and if you're thinking of getting into guitar you should probably invest in one. If you like to record into a DAW on a computer they will have a click feature built in too, and you can experiment with how fast or slow it clicks.

To start off with, set your metronome to 100 beats per minute (bpm). Then, using a single note, try to play every time there's a click. The key is to anticipate when the click sounds and play at exactly the same time. It may take you a while to get this right - so don't feel bad if it takes you 100 attempts. Tapping your foot along to the beat can also help you keep a rhythm.

Once you feel comfortable playing one note per click, try playing two notes per click. Once you feel comfortable with that, try this example (the tempo here is slower as it's more difficult to play).

Open the 'Playing a minor pentatonic scale with a metronome' tab (Right-click to download)

Tip: Play two notes of the scale for each click of the metronome. Make sure the notes are evenly spaced. Remember to alternate pick - or else!

Read more: Best metronomes

A good techno track can appear incredibly simple, but techno is not necessarily a simple genre to produce. From club-moving subs to subtly shifting percussion, there's a lot of nuance and skill that goes into creating a good techno track.

To help you improve your techno productions and inspire new ideas, we’ve compiled more than 40 minutes of essential techno tips, which you can watch in the video above, all drawn from our In The Studio series.

Between MusicRadar, Future Music and Computer Music, our team has been visiting electronic artists in their studios for decades. These are eight of our most insightful techno-focused sessions from our back catalogue.

How Tom Hades creates techno rumble

Belgian techno maestro Tom Hades has released tracks on labels including Drumcode, Suara and Rhythm Converted. Back in 2018, Computer Music visited him in his studio to watch him build a club-ready track from scratch, resulting in a session packed with techno production knowledge.

Particularly insightful is his approach to creating the genre’s distinctive low end ‘rumble’. The process starts with a solid four-to-the-floor kick, which doubles up as a sub bass using some creative effect processing.

He processes the kick with heavy saturation and compression, bounces it, and then sends it to a return track.

"Afterwards, what I do is I just put a lot of effects on the return track,” he explains. “An overdrive, a convolution reverb, a filter delay, and then I will finish with a simple filter, which gives me the typical techno rumble."

He then uses sidechaining tools to have the sub bass element duck out of the way of the original kick.

"This rumble is very nice because it's not really a character sound,” he says, “but it will give you the power of the kick underneath."

Matador’s drum layering

Irish DJ and producer Matador, aka Gavin Lynch, is known for his stripped-back, hardware-focused take on techno. Future Music visited him in 2015 to watch him break down the making of his track The Enemy featuring Felix Da Housecat.

In the video Lynch demonstrates his approach to creating layered, intricate percussion. He explains that, for him, the secret to thick-sounding drums is to layer multiple hits that occupy different elements of the frequency spectrum.

In his example a tight 909 kick is augmented by claps, hats and percussion lines sitting on top of one another. The clap, for example, is layered with a tambourine, while the hats are thickened with shakers.

To have everything sit together neatly, he runs the sounds through analogue-style processing, using EQ and compression. The effects are merely acting to gel everything together though, rather than reshape specific sounds.

"Instead of loads of EQs on one particular channel, I’ll layer up, say, two or three different hats and work on it that way,” he explains. “You get some original thickness there, so we're not depending on plugins."

Secret Cinema builds tracks with randomisation

From 1991’s hardcore sounds through to his more recent deeper, darker output, Jeroen Verheij has been a key figure of the Netherlands’ techno scene. His formidable discography includes releases on labels as diverse as Rotterdam, Cocoon and Drumcode, under pseudonyms including Meng Syndicate, Point Blank and his most well-known moniker, Secret Cinema.

In 2017, Computer Music magazine met up with Jeroen in his then recently-completed Amsterdam studio to find out how he used a sophisticated combination of software and hardware to create his tracks. In the video, he walks through the process of building a track up from its initial idea, into what would eventually become Ex-Drummer from his Seance EP.

As part of the creative process, he demonstrates how he uses the randomisation tools in Bitwig Studio to create spontaneous melodic ideas and percussion lines.

He does this by sending CV out to a modular synth, driven by a randomised LFO. A scale device is used to control the pitches being sent to the synth.

"If you're a drummer, you will always hit that snare drum on a different setting… which makes that one drum loop so interesting," he explains. “What I want to do is use all these little tricks with LFOs and random step sequencers to make those drum loops humanised again."

Paranoid London's studio jam strategies

When hardware-loving UK acid duo Paranoid London released their album Arseholes, Liars and Electronic Pioneers in 2024, Future Music magazine visited their London studio to catch up with band member Quinn Whalley, and find out how their proudly old-school, raw and analogue take on house and techno comes together.

As Whalley explained during the studio session, the band’s creative process is built around studio jamming and hands-on creativity. Key to making that approach work is having a handful of instruments and effects that the pair know really well – and sticking with that limited setup.

"I'm not a massive fan of using loads and loads of gear,” Whalley explains. “We don't really use a lot, each time we do a track we'll only have a couple of little things."

He also explains that, for Paranoid London, a lot of the magic happens when it’s least expected, and it’s important to capture everything.

"Record everything all the time, and keep it all,” he advises. "You can drag something that you did, like, three years ago, over whatever you’re doing now. When you take a bit of [something old] and put it over something that's in a different tuning or whatever, that's when weird stuff starts happening."

Luke Slater’s (Planetary Assault Systems) outboard effect processes

British techno legend Luke Slater is a man of many aliases. He’s probably best known for his work as Planetary Assault Systems, a moniker under which he’s been releasing tough, drum-machine-focused techno since the mid-’90s.

Slater’s prolific body of work also touches on ambient electronics, acid, house, psychedelic dance music and crossover electro-pop, working as aliases including The 7th Plain, Clementine and LSD, as well as under his own name.

In 2024, Future Music visited him in his hardware-packed studio as he prepared to release a new album under his L.B. Dub Corp alias via Dekmantel.

Among a whole host of production insights, he explained the important role that outboard effects play in shaping his sound.

"Most of this gear is actually made for rock and roll,” he explains. “But I just find it amazing when you put electronic music through it. You can get results that are sometimes amazing."

In the video he demonstrates how he uses things like Pultec EQs to shape his master track, the Overstayer compressor to apply overdrive and tone shaping, and a Culture Vulture to turn a 909 kick drum into a booming techno bass.

Creating rolling techno percussion with Wehbba

In 2017, Future Music paid a visit to Brazil-born techno producer Wehbba in his Barcelona studio, to break down the creation of his track Fake, released on Drumcode.

Tying the track together are lines of rolling percussion, which Wehbba explains are created using heavily processed drum machine rimshots.

"I really love rim shots, and I use them extensively in my production," he says. "This part was from a big jam on the TR‑8. I chopped up the bits I liked and picked different loops out of them.”

Wehbba then demonstrates how he adds body, character and movement to the percussion line. First he applies saturation, reverb and EQ. He then adds ValhallaDSP’s free FreqEcho to create an ensemble-like effect using frequency shifting.

Finally, he tames the sound using limiting and filters.

Less-is-more techno arrangement with BEC

Berlin-based techno producer BEC, real name Rebecca Godfrey, creates hypnotic, groove focused techno. Back in 2021, as she had just released her Pleasure Seeker on Pan-Pot’s Second State, Future Music visited her in the studio to find out how she created the trance-influenced track, Trusting the Mystery.

One particularly distinctive feature of the track is its simplicity.

"It's probably one of my most simple tracks ever,” Godfrey explains. “15 elements, and that's including three kicks, so there's actually 12 separate sounds in here. I think there's beauty in simplicity."

She demonstrates how the arrangement is built using a two-note melody and simple bassline, along with short, looping percussive patterns.

The tension comes from shifts in effects and automation, with reverb added to create space towards the drop, and the synth line’s filter cutoff opening as the track rises.

Tomy DeClerque's techno mastering tricks

In a classic 2012 studio session, Future Music visited Eastern European techno producer and DJ Tomy DeClerque in his studio.

One particularly noteworthy tip shared during that session is DeClerque’s approach to mastering. As the producer explains, he would often do a ‘mastering pass’ on a track at an early stage.

“Everybody else is doing the loop, then they create the song, then create the mix, and then they do the mastering,” he explains. “I prefer to do the mastering early in the stage."

This approach, he explains, helps him identify issues with the mix and production early on, that he can then fix.

"The mastering always brings up some frequencies you don't know,” he says, “so I prefer to do the mastering thing, break everything up, make it loud, and then go back and repair the things that mastering has brought out."

Any wood that comes into regular contact with foreign bodies needs to be kept clean, conditioned and protected – your guitar's fingerboard is no different. But I've read some conflicting advice when it comes to the best ways to look after different types of fretboard for acoustic and electric guitars – including rosewood, laurel, maple, pau ferro and ebony. It's time to simplify and demystify this essential piece of guitar maintenance.

As with a lot of things in the world of guitars, there isn't one correct answer. But we need to understand what substances perform different functions for our fretboards before we can experience the satisfaction of seeing and feeling them at their best.

Firstly, there are two main types of fingerboard wood; finish and unfinished. An unfinished fretboard has no protective lacquer over the wood and examples include rosewood, pau ferro, laurel, walnut and ebony.

Finished fretboards tend to be maple – though unsealed maple 'boards do exist. The finished type will only need cleaning and no conditioning because they have a thin lacquer that acts as the protective layer to keep things like sweat from seeping into the wood.

So before you condition a guitar's unfinished fingerboard, you need to clean it first. And a seemingly unlikely substance is good as a starting point.

1. Naphtha is a cleanser

Guitar techs first recommended Naphtha (petroleum distillate and most commonly referred to as lighter fluid) to me and I was initially confused. Surely a flammable substance and wood are a hazardous pairing?

It can clean finger oils out of your wood by breaking them down before it evaporates. This process will often make your unfinished dark fretboard wood look dry and even slightly opaque after application to work its magic. Don't worry – it's not a conditioner and this is supposed to happen. It's simply removing the foreign bodies in your 'board before you use an oil conditioner to rehydrate it.

It's also a safe body cleaner for nitrocellulose finishes when used sparingly – and like rubbing alcohol, it's good for removing residue from stickers that you might have regretted customising your guitar with.

2. Lemon oil isn't what you might think it is

Lemon oil is what comes up most frequently when the subject of fretboard care is mentioned. Some people even advise against using it. But if you are using it, is it a cleaner or a conditioner? Taylor Guitars co-founder Bob Taylor suggests it's both.

"I’d have no worries about using lemon oil on my fretboard," he stated in a Taylor Q&A with regards to acoustic guitars. "It’s safe. Use it only on the unfinished wood like the fretboard and bridge. The wood can dry out over time, and an oil like this, or linseed oil, or even mineral oil, can protect the wood and beautify it as well.

"Don’t overdo it," added Taylor. "Once a fretboard has been oiled a few times, you can slow down the frequency. The nice thing about lemon oil is that it cleans while it oils, so it won’t build up as easily, but be sparing. I don’t think your fretboard will need oiling more than twice a year, and eventually, once a year."

But a lot of the confusion around guitar lemon oil is due to assumptions about what it actually is. Despite the name, most guitar lemon oil is actualy often more mineral oil with less actual lemon oil than you'd expect. It's more for scent.

So it's actually mineral oil that's the more effective conditioner. This helps to break up the kind of unwanted oils from our hands, for the cloth to wipe away.

I like to do a clean, then add a little more on a second sweep over for a little conditioning – removing any excess with the cloth. And if you notice a guitar lemon oil that's actually yellow coloured, it's because artificial colouring has been used. This can actually help enhance your fretboard's "warm tone", as Gibson Master Luthier Jim DeCola points out in the video above.

3. Other alternatives are available for cleaning and conditioning

Aside from lemon oil, some of the big names in guitar accessories have their own takes on the ideal recipe for fretboard conditioner. MusicNomad F-One Oil is a cleaner and conditioner to be used on unfinished fretboards. It uses natural seed and tree oils and it smells nice too, I have to say!

Elsewhere, Daddario, Taylor, PRS and Dunlop all have their own takes on fretboard conditioning oil too.

4. Pick the right cloth for the job

In my article on how to clean your guitar, I pointed out why two different types of cloth are good for specific jobs when it comes to applying polishes and creams to the guitar. And this idea extends to the fretboard.

You only need a small dab of oil on each fret, then use a microfibre short pile lint-free cloth to move it around the surface of the fretboard without the cloth absorbing too much, too soon and before the oil can seep into the unfinished wood. Then use a different part of the cloth to remove any excess oil after a couple of minutes.

5. Don't forget your toothbrush!

Using your cloth to get into the edges of the frets and clean where grime can gather usually does the job, but for fingerboards that have been neglected for a while (and might be due a fret polish as the next step in a proper spring clean), a clean toothbrush can help with a little oil applied to it.

6. Maple needs love too

Finished (sealed) maple fingerboards don't need conditioning but are still due a clean every free months. Use your clean microfibre cloth with guitar cleaner to wipe away any grime from the 'board that may shorten the life of your strings.

I like to use MusicNomad's Guitar Detailer, because it can also be used on satin and matte body finishes too, plus its anti-static formula helps to repel dust and is has UV protectants to help sun damage for those summer days when your guitar is sitting on the stand.

• For more information on MusicNomad guitar maintenance and cleaning products, check out musicnomadcare.com, Thomann and Sweetwater.

When it comes to mixing a track, it’s difficult to insist that we adhere to a rulebook when we all know that, in the creative world of music-making, rules are there to be broken. We can learn a lot from past masters, and as Picasso once said, “Learn the rules like a pro, so you can break them like an artist.”

This is particularly applicable at the mix stage, where we decide where to place instruments and vocals in the stereo image.

Whether you are a DAW-user or rely on a hardware mixing desk, you can’t miss the ‘Pan’ pot, which is normally located immediately above each channel fader.

It might surprise you to learn that pan is short for panorama, and is used to dictate where a signal appears in the stereoscopic panorama. (Now you know why we tend to abbreviate the term to just ‘pan’!)

Generally speaking, we still live in a stereo world. The vast majority of us are lucky enough to have two ears, which presents the natural world around us as an ambient 3-dimensional image.

Translate this into production terms, and we are effectively trying to achieve a 3D audio soundstage, albeit with a 2D speaker setting. We do this by using a degree of smoke and mirrors.

The two control elements we have in our toolset are volume, which tends to suggest distance, and panning, which allows us to control the signal left and right in the stereo image.

If you have your stereo speaker system or studio monitors set up correctly, you should get a near-perfect soundstage, where both volume and panning play an intrinsic role. If you are a headphone user, the left/right position becomes even more obvious, as the speakers are in such close proximity to your ears, positioning becomes even more obvious.

We must insert a caveat here, because while most of us continue to work in stereo, there are numerous surround and 360˚ systems also available, and in this regard things work in a slightly different way. So, for now, we’ll focus on the basics, sticking with stereo.

When we first start exploring production techniques, there is a temptation to pan instruments all over the place and apply too much reverb. This will probably not yield the best results.

If you have ever listened to material from the 60s, you may have noticed that some examples of early stereo usage (by the likes of the Beatles, the Doors or even classic jazz albums) are extremely panned.

You might hear a group of instruments, such as drums, bass, and guitar, all grouped together on one side, while the opposite speaker may reproduce vocals or leads.

During the early days of recording, technical restrictions meant that instrumentation would be presented in this way. This hasn’t stopped some bands from replicating the sound in contemporary settings, even though it’s not ideal for headphone listening.

As recording technology became more refined, it became possible to mix across the whole stereo image, placing instruments in specific, less extreme, locations, and we can draw some useful tips from this approach.

The most basic rule of thumb is that the lower the instrument in pitch (frequency), the closer to the centre of the stereo image it should be placed.

Think of your mix as a tree; the main trunk is centred and buried in the ground, providing a solid foundation. These are your bass elements, but as the instruments rise in pitch (guitars, keyboards, horns), they act much like branches, extending out left and right.

For this reason, we generally tend to place the bass drum (kick) and bass instruments, in the very centre of the mix. If you are using a guitar and piano, you might offset these to a left and right 45˚ position.

It's quite likely that they may have complementary parts to play in your track, so splitting them up in this way will sound great in stereo, as well as affording them some space, so that the musical phrases don’t collide.

If you have two guitars and a piano, try placing the guitars left and right, with the piano more toward the centre. A lead vocal is just that, and will want to be fairly central, but there’s nothing to stop backing vocals being panned to provide choral space.

Some contemporary music can get a little chaotic, so if you find yourself with multiple synth parts, a good starting point is to think about where in the stereo image they appear, so that they don't occupy the same area and become less distinct.

Think of each sound or musical phrase as having its own specific point in the overall stereo picture you want to paint, which will allow for more effective sonic separation.

Drums kits operate a little differently, because it’s a unit which contains a number of constituent elements.

Many drum machines or drum plugins will take care of the panning for you, but the basic guidelines suggest that we centre the kick and snare, while placing the hi-hat slightly off-centre. This is often placed to the left, to mirror a conventional drum kit setup, for a right-handed player. The ride cymbal would tend to be panned to the opposite side, with toms (normally three) moving from left to right, for each tom, high to low.

If you are lucky enough to have a real live kit drum recording setup, be aware that we tend to use a stereo pair of overhead microphones, alongside individual microphones for each drum.

It's important to ensure that the panning of the individual signals match up with the imaging of the overhead stereo mics, otherwise your result will sound pretty strange!

Dealing with orchestral instruments can present its challenges too. A useful tip is always to consider where an instrument sits, when it plays within a group. With an ensemble of around 80 players, a full symphony orchestra presents the biggest challenge of all, and also contradicts some of our tried and tested formulas…

Let's consider the string section; from left to right, we hear 1st violins, 2nd violins, violas, cellos and double basses. You will have noticed that the frequency assignment moves from high pitches on the left, to low pitches on the right.

Despite the fact that the basses are not in the centre, it will still sound impressive when panned. If paired with instruments from a band environment, your mix will feel nicely grounded, thanks to our previously mentioned conventions.

It can be very tempting to pan instruments all the way to the left and right, or ‘hard panning’ as it's known.

This can sometimes work well, if double-tracking instruments like guitars and keyboards, but in most scenarios it is just too extreme.

Always be objective - it might sound great, but if it doesn’t, adjust the panning inward slightly, away from the extremities.

As you are effectively using panning to create a mix space, you can create some wonderful effects using panning.

You can try automating panning in your DAW, so that appropriate sounds or instruments move from one side of the stereo image to the other. Alternatively, try placing an instrument to one side of your mix, with effects such as delay or reverb positioned on the opposite side. This can be particularly effective with electronic instrumentation.

Finally, it is always worth checking back your mixes in mono. Professional engineers tend to have a single speaker on hand, so that they can check that the stereo mix condenses and sums down to mono without phasing issues.

You can achieve the same thing in a DAW, by switching your master output to mono.

Despite our love of stereo, there are still lots of radios and phone speakers out there that operate in mono. We’re not predicting that mono will become the new stereo, but you never know - cassettes and vinyl are back, so why not mono?

Here's a more in-depth video guide to panning using Logic Pro as our DAW of choice

The very term ‘saturation’ sometimes feels like it has a negative value attached to it. Throughout recording history, technology has sought to deliver the cleanest signal possible, so why for the love of Joe Meek would you want to turn a clean signal into a distorted one?

It turns out, the answer is relatively simple. As human beings living in an organic world, we quite like impurity and a measure of chaos - and that's where sensible use of saturation can bring an awful lot to the musical table, whether it's individual instrumentation, as well as an entire mix.

The term distortion is very closely linked to saturation, but while they are subtly different, one could not have existed without the other.

If we go back around 100 years, the predominant music of the era was jazz, and particularly big bands. Electric guitar amplification was not particularly advanced, and would now be probably described as ‘lo-fi’.

Guitarists would often turn up their equipment as high as they could go (we don’t think ’11’ was a thing back then), in an attempt to compete with the loud acoustics of a large band. This would lead to equipment overdriving or speaker cones failing, with the resulting sound being something close to the guitar distortion that we know today.

With the advance of electronics, distortion pedals began appearing in 1962, and by the 70s, they were both commonplace and an essential part of a guitarist’s rig. This gave guitarists ultimate control over their distorted sound, without the need to destroy an amplifier on a weekly basis!

But why should guitarists have all the fun? With the rise of synthesisers in the 70s and 80s, synthesists and producers realised that they could get some pretty gnarly tones by attaching the same distortion pedals to synthesizers.

Nothing exemplifies this more than the acid house era of dance music, from the late 80s and early 90s.

It was this period where the now-legendary Roland TB-303 was repurposed from a humble bassline machine to the must-have device for the genre. It turns out, whacking up the resonance on the 303 filter produced an amazing sound through a distortion pedal. Club-avoiding guitarists, however, were largely non-plussed

So what exactly is saturation, and what does it bring to the production table?

By and large, saturation is designed to be more subtle than distortion, often with a penchant for mimicking good old-fashioned analog equipment. You remember our previous point about clean signals? We rest our case!

Saturation excites and distorts certain frequencies and harmonics in the signal, depending on your chosen plugin. What's more, we can trace the saturation story back to the late 90s.

Back then, CDs were the predominant format, and many musicians and producers were reporting that the overall sound of the digital signal chain was a little cold and sterile.

Couple this with the emergence of one of the first rack-mounted units to offer a one-stop mastering solution - the TC Electronics Finalizer - and basic mastering saturation became commonplace.

TC Electronics described this saturation as 'Digital Radiance Generation', or DRG for short.

DRG applied second harmonic distortion to a mix, designed to mimic analog equipment and specifically analog tape, so it's no surprise that it was quite a hit.

However, over-using it could quickly destroy a mix, forcing mix engineers to pay close attention.

Fast forward to the current DAW age, and we are now flush with a plethora of different saturation plugins, with many being very highly regarded and offering bespoke solutions, depending on your production requirements.

One of the big boons of plugins, is that you can use them as many times as you like, and while it is probably a bad idea to sprinkle saturation on every instrument in your mix, there are definitely places where employment can pay dividends.

If you find yourself recording an electric bass part directly via your audio interface, you could choose to use amplifier simulators to colour the signal. This could be regarded as a form of saturation, but you could also go all-in and just apply a saturation plugin to warm up the signal.

One of the most highly regarded in the current market is Decapitator from Soundtoys. Not only is it flexible, but it can go from incredibly subtle to total extreme very quickly.

When applying saturation, it is always really important to listen objectively. The moment you feel the signal is getting too aggressive, it's time to back off, and that may not necessarily be exclusively down to the saturation amount control.

Just about all plugins will be equipped with a Wet/Dry see-saw control, and adjusting this can scale an extremely saturated sound to help it blend in more nicely.

Of course, this principle could also apply to sampled bass sounds, which may be a little too clean for your mix. It's also a concept which could be applied to drums, specifically if you are working with samples within your DAW.

It's a common complaint that many drum samples are just too clean, so it could be very useful to add a little saturated colour to your drum track. The only problem is, the drums themselves often sound great with saturation, but the hi-hat and cymbals can react less favourably…

A sensible solution requires the creation of 2 drum tracks, both with accompanying MIDI parts. You can then mute all of the drums on one channel, muting all of the metallic instruments on the other. This leaves the path clear to saturating the drum elements on their own, with no effect on the cymbals and hats.

Before you get lost in instrument saturation heaven (or hell, if that's your preference) be aware that some instruments do not embrace saturation so readily, particularly instruments with a more acoustic persuasion, such as acoustic pianos, string sections or brass and woodwinds. There's no harm in trying, but don't be surprised if saturation does little to improve the sound.

With a slight sense of history repeating itself, we have to revert to TC’s DRG concept, and discuss how beneficial subtle saturation can be to an entire mix.

Mastering suites, such as iZotope’s Ozone, offer a number of saturation solutions, designed to be used during the mastering phase. These include tools to excite upper frequencies, and elements for inducing the sound of vintage tape.

But if you want a little more control, one company that personifies the application of saturation to mimic analog is Universal Audio. It has produced an array of exemplary plugins which are perfect for subtle saturation across your mix.

If you are nervous about over-applying saturation, take a look at the Verve Analog Machines plugin, which is available in two forms; the first is an entry-level Essentials version, offering four saturation models, while the upper-level version provides 10 models.

The idea is simple - apply across your mix, and adjust the tone and amount controls to suit your production. The included models range drastically from subtle, through to downright dirty, so it's important to be objective.

For even more control, UA’s emulation of the Ampex ATR-102 Mastering Tape Recorder is stunning. This plugin can induce everything from saturation through to hiss and even wow and flutter. It just depends how extreme you want to go!

One final thought. We've mentioned on several occasions that you need to be objective about your saturation exploits, and be careful not to overuse. You can easily destroy a mix by being a little overzealous.

If you find yourself regularly creating or delivering stems of your mix, be aware that many saturation plugins induce noise.

If you deliver half a dozen stems and they all have noise on them, when all the stems are played back together, that's going to be quite a racket. It’s probably best to defeat the noise element as much as you can, in this setting, so that your music doesn't get drowned out by hiss.

We spent many years trying to get rid of tape hiss and noise, so you don't want to end up back where we all started.

At the dawn of electronic music, as the experimental nature of a new-fangled instrument called the synthesizer was in the ascent, musicians that embraced this new technology were confronted by a problem; how to describe these new sounds?

As the polymaths at the BBC Radiophonic Workshop, were acquiring a reputation for creating new and interesting sounds (initially using tape-based methods, before migrating toward synthesizers) the call for a form of cataloguing system was essential.

Being the BBC department that supplied sounds for shows such as Doctor Who and Blake’s 7, one sound was becoming particularly commonplace - the ‘Laser Zap!’

The BBC didn’t have the monopoly on these sounds though. Star Trek also featured many-a-zap, with subtly different ‘zaps’ being required, depending on your phaser setting, or species of alien…obviously!

It didn't take long for electronic artists to embrace and mutate the idea of the zap, and use it in a musical context.

You could employ a laser zap with a long sweep, as a form of electronic cymbal crash. But, if you sharpened up the decay of the envelope, it suddenly became a heavy duty sound, with snare or electronic hi-hat potential.

That’s certainly what Kraftwerk thought in 1981, when the track Numbers (from Computer World) made its inspirational splash.

The insistent use of Hi Q, in a form of stereo hi-hat battle became a much copied production detail, influencing hip-hop, R&B and rap, as demonstrated on Planet Rock by Afrika Bambaataa and the Soul Sonic Force, and 1, 2 Step by Ciara ft Missy Elliot.

But how did the zap migrate from its association with Klingons, to its more common description of Hi Q? That's all down to the way we create the sound.

There are a couple of ways of acquiring this sound, but the most notable version, which we are going to recreate here, relies explicitly on the filter section of a synth, particularly the resonance, which is also described in some circle circles as Emphasis or just plain ‘Q’.

When dealing with a filter, there are two main components which we tend to regularly control. The first is the cutoff frequency, which in the context of a low pass filter (which is what we will need here) is the element that filters out upper harmonic content from a wave form, such as a saw or square wave. This allows the lower frequencies to pass through unfiltered. The point of frequency where this filtering occurs, is dependent upon the position of the cutoff fader or pot.

The resonance control, which normally resides alongside the cutoff, creates a form of accentuation and feedback, which in many filter designs results in a form of self-oscillation. This pure sinusoidal sound, is what we are after. Or, to put it another way, set our resonance control to a Hi Q.

But the story doesn't end there - the sweeping sound is generated using modulation.

Ideally, we will need a synthesizer which provides us with two envelopes. Just about all synths that offer two envelopes will default to one envelope for amplitude/volume, with the second envelope modulating the filter cutoff. It's this second envelope, and particularly the decay phase, that is going to do the heavy Hi-Q lifting.

The even better news is that we can adjust the decay phase to seamlessly move from electronic cymbal crash or laser blast, to a sharp Hi Q. Hence, it pays to choose your host synthesizer well. Most software synths will be able to do a passing facsimile, but any synth with snappy envelopes, and an aggressive filter, is going to do the best job.

In vintage hardware, several synthesisers offered the perfect Hi Q potential, with the SCI Pro-1 and ARP Odyssey being stylish contenders. The popularity of the Minimoog, both then and now, makes it the ideal partner.

We are going to use Cherry Audio’s Miniverse software plugin for our example, but most plugin synths will offer the capacity to create something very similar.

Step 1: As our Hi Q is entirely dependent upon the filter, we can start by switching off the oscillators. If the synth you are using does not employ switches, just turn the volume of the oscillator(s) down, or select a wave such as white noise, as this will be less prominent within the overall sound.

Step 2: Next, we move over to the filter section, where we need to shift the cutoff frequency to a relatively low position. The exact position of the cutoff pot or fader will be entirely dependent upon the synth you are using, so try placing it in the lower 25% and this will offer a starting point. At the same time, increase the resonance/emphasis/Q to its maximum.

Step 3: We need to set the amount of modulation, which is being applied to the cutoff, from the filter-assigned envelope. Locate the relevant pot or fader, and set to a reasonably high value, possibly the maximum.

Step 4: Adjust the filter envelope settings, to the quickest attack available, with a very short decay. Remember, the decay time is the element that will dictate the snappiness of the sound. If it's long, we are into laser zaps again, but keep it short, and we have a patch that can be used rhythmically. It is also essential to place the sustain level at zero.

Once you have applied the settings to the envelope, you will probably need to revisit the cutoff frequency and modulation amount controls. Trigger your sound from a pad or keyboard, and adjust these elements to suit your sound and track.

Step 5: Depending on the synth you are using, you may find that a very low pitched hum occurs, at the end of the sound, if you play and hold. This is the sound of the filter, continuing to be audible at a very low frequency/pitch.

It is easy to eliminate, with the use of the amplitude envelope; make sure that the sustain control is turned all the way down, and play around with your attack and decay times, to get the best result. They will need to be in a similar position to the filter envelope’s attack and decay phases.

Now you have your sound, it's time to place it in your track, and there are many production settings where this can be deployed.

If you want to think of your Hi Q as more as a hi-hat sound, simply play a hi-hat rhythm, and it will quickly add some momentum to your track. If you want to go full German techno, use two versions of the same sound, with one panned hard-left, and the other hard-right. You can also use a stereo delay, to get similar results.

You can also change the decay settings to mimic sounds more like an electronic snare, or even lower the cutoff frequency control, to create a superb electronic kick.

Of course, you can also apply many different style of effect, and Hi Qs absolutely love a bit of overdrive, so try dirtying up your signal to add real aggression.

Influential songs come in many shapes and styles, but making a track which goes on to universally transform music and culture is a rare thing indeed.

Giorgio Moroder and Pete Bellotte produced a number of hits with Donna Summer, including the sublime Love to Love You Baby in 1975.

In the following year, while working on her fifth album, I Remember Yesterday, the production team decided that they wanted each track to reflect the musical flavours of various different historical decades.

They explored 60s girl groups and navigated 70s disco, but when it came to the last track on the album, they wanted to create something that looked ahead to a hypothetical, electronic-tinged future…

The result was the epic I Feel Love, which charted at number one in the UK for four weeks during the summer of 1977.

While its commercial impact globally was equally impressive, I Feel Love’s true legacy would be how it became a benchmark for electronic music producers.

Alongside pioneers such as Kraftwerk, the track acted as a green light for musicians and producers who wanted to produce music entirely electronically. So, by wildly envisioning the future, Moroder, Bellotte and Summer essentially invented it.

In fact - as recounted by David Buckley in his Kraftwerk biography Publikation, when Brian Eno (then working with David Bowie on his hailed Berlin trilogy) first heard the record, he raced to tell his creative accomplice.

"One of the songs which certainly impacted greatly in the summer of 1977 was a song which sounded as if Kraftwerk had gone potty and recruited a bona fide American soul singer,” wrote Buckley. “In fact, it wasn’t Kraftwerk, but Italian musician and producer Giorgio Moroder. ‘One day in Berlin,’ says Bowie, ‘Eno came running in and said, “I have heard the sound of the future.” He puts on I Feel Love, by Donna Summer, he said, “This is it, look no further. This single is going to change the sound of club music for the next 15 years.”’ Which was more or less right.”

Many years later, the technical commonalities between this track and the explosion of electronic dance music, became impossible to ignore - and I Feel Love became regarded as a vital part of its birth. You can read more on that aspect of the song over in this article.

It was arguably the first high profile track where a sequentially looped bassline was coupled with a four-to-the-floor bass drum, and electronically replicated off-beat hi-hats.

Giorgio Moroder explained that when writing the song, the bassline was the first thing to be composed.

He had borrowed a Moog Modular synthesizer from classical composer, Eberhard Schoener. Aided by Schoener’s assistant, Robby Wedel, they set about synchronising the Moog’s step sequencer to a click track, recording to 16-track analog tape.

This was not without its difficulties and complexities, as the Moog would continually wander in pitch. It was something of a happy accident when Moroder’s engineer accidentally added delay to the bass part, and cementing the infectious pattern that we know today.

Thankfully, we don't need a room taken up by a Moog Modular (or a huge reel-to-reel tape machine) to recreate this classic bass sound. In fact, it's relatively easy to build in our DAWs using a synth plugin (and a little delay).

In order to capture the sound accurately, we will need synthesiser with at least two oscillators, or possibly even three. If using a two oscillator sound source, having the ability to modulate at least one of the waveforms (PWM) would be an advantage. However, we are choosing to adopt a Minimoog oscillator sound source, which will offer us a very similar sound up-front, but will require a slight workaround.

As the original sound was created using a Moog Modular, we are going to use the Minimoog plugin from Universal Audio, which gives us the initial sound, but like the original Minimoog, it does not offer PWM. Thankfully we have options, which we will explore while building this patch.

As we will need to use tempo sync’d delay later on, it is easiest to load your synth via your DAW, rather than as a standalone instrument. Be sure to load a mono instance of the instrument. Once loaded, create a default patch and head to Oscillator 1. Set the octave range to 16’ and the waveform to a sawtooth.

Make sure the oscillator is turned up to its maximum level.

Move to oscillators 2 & 3, and set both ranges to 16’. Be sure that Oscillator 3 has ‘OSC 3 Control’ switched on, which means that its pitch is controlled from your keyboard or MIDI.

As we do not have PWM available on the Minimoog, we are going to add some different harmonic interest using a square wave and modulated pulse waveform, which are both available with Oscillators 2 and 3.

Once you have selected your waves, make sure that the volume of both oscillators is turned up to maximum.

If you are using a synth which allows PWM, use a second oscillator set to a pulse wave with PWM modulated by the decay phase of an envelope. If you listen carefully to the original, you can hear the interesting harmonic content, which is likely generated by the summing of a considerable number of oscillators within a modular, coupled with movement created via PWM.

We have employed a subtle workaround, using fixed waveforms to offer a similar sound and spirit.

There is a fine-tune option available with these oscillators too. We do not want to detune very far from Oscillator 1, so adjust the tuning so that they are all fairly close together in pitch.

Move across to the filter section and set the cutoff frequency to around 30%. We don't want anything too bright, particularly as we want to add some envelope modulation, which we do by adjusting the Amount of Contour pot, to a value of 4 (or around 40% if using a different synth.)

Moving to the envelopes (or Contour Generators, in Moog language) we need to make adjustments to both the filter and loudness (amplitude) envelopes. Firstly, make sure that the decay switch is activated, then reduce the attack pots to 0. Both sustain level pots, should be reduced to 0 as well.

Finally, adjust the decay pots, so that the filter decay pot is set a little bit higher than the volume decay pot, somewhere between 200-250 milliseconds. If you don't have a timing scale available, these both need to be fairly snappy, so keep the decay times relatively short.

Now we have our synth sound set up and ready to go, you can play around and try replicating the original I Feel Love sequence, or create something compositionally of your own. Either way, set your DAW’s tempo to around 126 BPM. The original track clocks at 126.5 BPM, but you don't have to be that exacting!

To replicate the delay aspect of the sound, route a Bus or Aux from the initial bass channel, ensuring that the Bus/Aux channel is also in mono.

Send a healthy amount of signal to the Bus, and add some form of delay. In this setting, the simplest delay possible will be the most successful, as we do not want stereo or tape regeneration effects. If your delay offers a wet/dry control, ensure that it is set to fully wet.

We have opted to use a simple timing/sample delay, and have set our timing to match our tempo of 126.5 BPM.

You can equally use a sync’d delay set to 16th notes.

Once assigned, pan the original bass toward the left, and the delayed signal equally to the right. If necessary, adjust the volumes of both channels so that they match. Moroder discovered that hard panning these channels was too extreme, particularly for playing in a club, so a revised mix was set with moderate panning left and right.

You should now hear each bass note replicated through the delay, creating a ping-pong effect, lending itself to the repetitive pattern that we know from the track.

If you want to draw further inspiration from the original, you can hear that there are lots of embellishments to the bass part, which include accenting the first couple of beats with different instrumentation.

You could also try adding a sub bass, which is probably easiest to do by lining up another synth, copying the MIDI from your bass track, and using a heavily filtered sound to play an octave lower.

Beyond that, add a string machine with plenty of rich phasing, and you too could be on the way to creating an electro-pop classic.

If you've ever visited a professional recording studio, you may have noticed the large amount of outboard equipment that many studios still employ. Nestling alongside a traditional mixing console, there will probably be an abundance of outboard rack units, offering a host of external processing. It’s likely that many of these will be compressors.

There was a time when compressors were only available in hardware, meaning that you would need a single compressor per track - which could feasibly add up to a large number of units.

Today, as we’re operating largely in the plugin-heavy DAW-domain, the landscape has changed significantly. No longer do you have to have an external compressor (although many of us still do) and now you can use as many as you like from within your storehouse of plugins and emulations.

First and foremost, compression is all about dynamic range. The vast majority of compressors offer the same basic set of controls. Some compressors extend a little further, but a basic set of half a dozen controls will likely always be available.

But why is it necessary to compress? Let's suggest that you are working within a conventional song style, with a basic band lineup of drums, bass guitar, guitar, and vocals. While all of these elements would benefit from degrees of compression, there are two elements that will benefit above all others.

Firstly, the vocal part will probably want to sit dynamically above the band elements, and secondly, the bass part provides the perfect foundation for a traditional song format. There is no better way to accentuate this foundation than applying a touch of compression to the bass.

So let’s concentrate specifically on the bass, focussing on appropriate settings and styles of compressor that bring something to the bass party.

Most compressors, whether hardware, software or in pedal-form, typically present some form of metering, either in the style of LEDs or as a needle on a meter.

Despite these being analog elements, it is very common to see a VU meter facsimile on a plugin compressor.

So, let's go through it step-by-step, assuming no prior knowledge.

Apply your chosen compressor plugin to your bass channel (we're using Logic's in-built compressor) and play your track.

The chances are, the compressor metering will not be doing very much at this stage. This is because we need to adjust the control known as Threshold.

Think of this control as being a monitor for the level of the signal entering the compressor circuit. The level has to reach the point of threshold, before the compressor begins to work.

Consequently, reducing the threshold level, should elicit a twitch of the metering, at which point the compressor will begin to do its thing.

Increasing the threshold will probably prevent the compressor from doing anything, unless it is exposed to a very high level of signal.

As a starting point, decrease the threshold point so that the metering starts to twitch.

The meter will probably be displaying something known as Gain Reduction - there may even be a switch to activate this mode.

The greater the movement of the meter, the harder the compressor is beginning to work, but what is it actually doing?

Once your bass signal has extended past the threshold point, it begins to limit the signal.

We can control the amount of compression, which is dictated by the Ratio control.

Working in tandem with the threshold, an increase in the amount of ratio will dictate a greater level of compression, lowering what would be the level of output if the compressor were not in place.

If you are working with some form of synth-bass, in a style such as EDM, you will want a relatively low threshold point, with a relatively high ratio.

This will create a bass track which is as flat as a slab of concrete and the perfect foundation for a commercial EDM track - quick to take effect and heavily compressed!

If, on the other hand, you are working in a more acoustic environment, you may want to increase the level of threshold, and reduce the ratio of compression.

This will elicit a more natural response, which for an acoustic setting is exactly what you want. You don't particularly want to hear what the compressor is doing, you just want to keep your bass under control should the odd rogue peak emerge in the bass part.

You may also find a control described as Make-Up Gain; think of this as a secondary volume control, which allows you to increase or decrease the volume outputted, post compressor circuit. This can be particularly useful - especially for very heavily-compressed bass parts - when you want to balance against a drum part.

There are three other controls that will often present themselves in a compressor environment; firstly, the Knee control dictates how subtly or heavily the compression will take effect at the point of threshold.

A subtler iteration is described as soft-knee compression, although you can increase the Knee control to generate what's known as hard-knee compression. The latter being particularly useful for more electronic forms where more literal application is required.

You should also see Attack and Release controls; the Attack control is used to set the time that it takes for the compression to take effect once the signal reaches the threshold.

Setting this value too low can sound very unnatural, but offset by just a few milliseconds, and it will definitely have a positive effect in most instances.

Similarly, the Release control dictates how long the compression will take to subside. This can be particularly useful for pad-like basses, which might have a prolonged sustain at the end of each note.

You would be forgiven for thinking that all compressors are equal. Unfortunately, they really aren't, as alongside their basic compression duties, they often have a sound of their own.

The 1176 FET compressor, is a particularly agile and impressive unit and plugin for keeping control of the rapid transients that can occur in bass sounds, particularly electric guitar and double bass. This front-end-attack agility will keep any fast bass parts under control, while also colouring the bass signal in a really pleasing way.

Similarly, the LA-2 Compressor is another studio legend which lends itself to bass frequencies. It is well known for being slower off the mark when it comes to handling quick transients.

Hence, it has become very commonplace to use these two compressors in tandem, with the 1176 first, followed by the LA-2.

It's the perfect pairing, and is available in numerous forms virtually from many plugin companies, particularly its architect, Universal Audio. It has a lovely virtual version of both the 1176 and the LA-2A.

You will also note that many included DAW compressors adopt similar descriptions in an attempt to clone these hardware staples.

As with all signal processing, there is no substitute for taking your eyes away from the screen - and using your ears to ascertain what’s needed. As it differs track-to-track.

Zone in on the type of bass sound that you want, and if necessary, don't be afraid to add a back-end EQ after the compression part of the signal chain.

With so much beautiful colouration from the likes of the 1176, it is possible to overcook a signal, and the added bonus of backend EQ, will provide a useful level of control for easing a too-present bottom-end.

In commercial music, few relationships are as fundamental to the success of a track as the bass and kick drum. The bass, whether acoustic, electric or synthetic, provides a harmonic foundation, as well as a stylistic feel of funk, house, EDM, or good old rock 'n' roll! The kick drum (aka the bass drum) supplies pulse, impact, and completely reinforces the bass’s groove. Whether it’s 4-to-the-floor or IDM, arming yourself with a few simple tricks could reap enormous rewards.

This unique association reaches right back to ancient civilisation.

European, Native American and African cultures all employed a form of stomping and singing, as a form of communication and celebration. Commercial music simply distills this primal music-making into a produced format. Have you ever felt the bass and kick from a large sound system hit you in the chest? Stomping and singing has the same effect - it was just a little quieter back then.

It's relatively easy to build on this concept, using technology to fine-tune the principle. The starting point should really be considering the style of music that you are working in. We live in an age where music is (rightly or wrongly) categorised, but you can certainly learn from past masters.

Kick and bass will be integral bedfellows for most electronically-oriented styles of music, such as EDM, House, Techno and Trap. They all employ the most simplistic construct, where a repetitive kick pattern tends to be used.

This isn’t an exclusive observation, but it is a fixed hallmark of the electronic domain, where looping and repetition is paramount.

When writing or producing tracks in these styles, it is probably most useful to begin with a bass line. Once you have your line composed and recorded within your DAW, the kick can be used to accentuate and highlight specific notes from the bass part. There are a couple of ways to explore this; firstly, you may feel that you want to keep it simple, and use the kick on each beat of the bar.

This is the traditional 4-on-the-floor concept which dates back to the 70s. In fact, it harks back to the age of disco. There has been little change, except for the nature of the sounds and the punch that they can provide, and it is this ‘punch’ element that is so pivotal here.

If you listen, you will hopefully note that where a bass and kick note coincide, you get a particularly strong sense of accentuation.

This is because there are two transients occurring at the same time, so it stands to reason that it's going to sound impactful and epic.

If you are seeking inspiration for the effectiveness of this, listen no further than the classic Chic track, Good Times. Regarded as one of the most recognisable bass lines of all times, you can hear the fusion very clearly.

The first three notes in the bass part are doubled by the kick, on just the first and third notes. This sense of accentuation provides a sense of musical phrasing.

Fig 1. The accented 1st and 3rd beats from the bass line of Good Times

It’s a similar association with a far more modern production, 24K Magic by Bruno Mars. The bass and kick collide in perfect synchronisation for that extra impact on beats 1 & 3.

A second approach to exploring this relationship, employs less synchronisation, and more syncopation. It's certainly true that the relationship between these two instruments is more fluid in styles of music such as funk or jazz, but there are other idioms that play against the accentuation concept.

Italian producer Robert Miles, created a form of music which became known as ‘Dream House’ - best defined by his enormous 90s hit, Children.

This entirely electronic genre places the kick on every beat, while a relatively static bass part plays on the 1/8th notes, in between each main beat. This placement could also accurately be described as an offbeat.

Fig 2. The pulsing 1/8th note pattern between kick and bass synth from Children. The synth bass notes end exactly in time with the start of each kick strike

Other commercial styles, such as Drum and Bass, often treat the kick/bass relationship a little bit differently. Many drum loops used within this idiom are tuned to a higher pitch, along with the increased tempo. This places the kick in a higher frequency domain, making space in the mix for the bass part.

Once you have written your bass and kick components, you can start to concentrate on how they blend together at the mix stage.

One of the primary things to consider is the timing of each element. If you are working with a heavily-quantized kick part, it can be worth examining the attack phase of your bass sound. If it is an electronically-produced sound from a synthesizer, it can sometimes sound more effective with a slightly elongated attack phase, accessed via the amplitude envelope.

This is dependent upon your chosen synth, as not all attack times are equal! Even a slightly extended attack time can make the combination of the two instruments sound more organic and effective. If you are working with a live bass or bass guitar sample, try delaying the track by a few ticks, as this may help the bass part to sit more eloquently with the kick.

Our ear tends to hear heavy transients first, so a slightly delayed pitch element (provided by a bass part) should sound great, just a few ticks or milliseconds later.

If working with a synth-bass sound, you could also try using a secondary envelope, to shape filter cutoff. You’ll want a quick attack, followed by a quick-ish decay, which will generate a brightness which plays beautifully against the dull thud of a kick. Adjust your decay phase timing to suit your sound and track style.

Many mix engineers work with the drum and bass channels as a starting point for a track mix. There is a rationale here, which naturally flows from the relationship between these two instrumental elements.

If you can get the perfect bass and kick (and drum) parts to work together, you'll have an incredibly solid foundation to build upon for your additional mix elements. But for our money, it all starts with the composition and placement of beats.

We have become so accustomed to our DAW tools that we often don't stop to consider the positive implications of not using them.

Let’s talk about quantize. For the uninitiated, quantize is the semi-automatic process of locking anything you record into your DAW to a pre-ordained timing grid, meaning that all of your tracks snap together in perfect synchronisation with your set BPM.

Sounds ideal, doesn't it? And while many forms of commercial music rely heavily on the fixed rigidity of quantize, the benefits of not using this operation can be overtly positive too, particularly as AI continues to infiltrate music streaming platforms. Today, listeners are more reassured when they subtly recognise those all-too human qualities in the music they listen to.

The argument for not pressing the quantize button begins with the style of music you want to create.

If your production world revolves around EDM or other electronic forms of production, the sound of heavily-quantized beats, emulating the sort of thing you'd expect from a retro drum machine, is part and parcel of your production lexicon. But that's not to say your other instrumental elements can't be loosened a little.

If your music contains live (or sampled-live) instrumentation - from a topline singer down to a bottom-line bass - there are many advantages to not leaning on quantize - but there is a caveat!

If you're tracking the part yourself, this may mean that you need to attempt to record a part several times in pursuit of the perfect take.

It's the contemporary equivalent of how musicians used to work when recording to analog tape. It's also why good musicians, who have spent many years learning and practicing their craft, tend to attract attention and command generous fees.

To understand the benefits of this undeniably more labour-intensive way of working, we have to ask what it is that we are doing when we click the quantize button?

Take a strummed acoustic guitar; with six strings, it's impossible for all six notes to be plucked at exactly the same time. This becomes even more exaggerated when using a plectrum.

Depending on the upward or downward direction of travel for the plectrum, we will either hear a top or bottom note first, followed by the other notes in the chord.

Admittedly, these notes may only be spread out over a few milliseconds, but our human ear perceives this as a pleasant imperfection - and we like it.

It's the exact same rationale for a pianist playing a chord on a piano.

Even if all the notes are attempted to be played simultaneously, there are still likely to be subtle discrepancies between the timing of each of the notes.

As pianos and some keyboard sounds have a heavy transient at the front of each played note, the staggered timing effect softens the sound that we hear, and just as before, some part of us recognises these failings as a by-product of being human.

You can apply this same ethos to any polyphonic instrument or group of instruments, from horn and sting sections, through to conventional drum kits.

We already mentioned that creating the perfect recording may take a little practice. It could be that your playing skills need some polish, but the good news is there are other ways to approach this concept, using tools within your DAW.

Next time you play a keyboard part, see how it sounds in an unquantized state. It's fair to say that once the light goes red and we start to record, we often find ourselves panicking, and rushing against the DAW metronome.

But, leave your track unquantized and try delaying it by a few ticks or milliseconds, but be sure to listen to your playback with objective ears.

It can be a subtle alteration, but one that can have a significant effect on the sum-of-parts. It’s also just as applicable to recording audio as it is to MIDI.

1. Try adding a small delay to an unquantized part, to improve playback timing.

If the delay approach does not yield the results you hoped for, there is another option that can be exceptionally useful for less experienced players working within the MIDI realm.

Record your part and quantize your recording. Then locate the ‘humanize’ function within your DAW.

Most humanize control panes will provide the option to assign the degree of 'humanize' you wish to apply. This will be dictated by the tempo of your track, as a slower track will require less humanization.

As a starting point, try applying a humanize level of around 10 ticks.

This equates to each note being nudged earlier or later in a randomised fashion, by up to 10 ticks in either direction. You recall our previous pianist analogy? This operation will yield a similar effect.

2. A typical quantized piano part, tight to the grid.

3. Adding small amounts of humanize will make a MIDI performance sound more authentic.

Now… listen!

It’s important to listen back to your work with analytical ears. Really zone in on the part that you want to affect and try different levels of delay or humanization to tilt the sound of your chords to being as unmistakably real-sounding as possible.

With the release of Live 12, Ableton added a host of fascinating MIDI tools to the DAW that introduce new ways to inspire melodic ideas. These are particularly useful for creating quick basslines, melodies and riffs to use within a track.

In our latest video tutorial, Ableton Certified Trainer Tom Glendinning explores a variety of ways to create and develop monophonic patterns in Live. In the video we’re focusing on basslines, although these techniques can all equally be used for melodic lines by simply changing the octave range and sound being used.

Seed MIDI generator

With the launch of Live 12, Ableton added a selection of MIDI generator tools designed to automatically create patterns from scratch within MIDI clips. Each MIDI tool specialises in a different form of programming. Rhythm mode leans towards drum and percussive programming, while Stacks is designed to generate chord shapes.

When it comes to generating ideas for bass and lead lines, the Seed MIDI tool is particularly useful. This uses a few core settings to control the MIDI notes being generated. The first of these is the pitch parameter, which is used to control the range of notes generated. The second is duration, which controls the range of timing and length of notes. Velocity sets the maximum and minimum velocity levels of the MIDI notes.

Seed also has a voices parameter, which alters the amount of simultaneous notes being generated and adjusts the harmonic complexity of any chords. Since we’re looking to create bass or lead lines, which tend to be monophonic, we set this to just one voice. Finally, the density parameter controls how densely or sparsely spread out the resulting notes are.

With auto mode engaged, Live will instantly create a MIDI pattern and the parameters can be adjusted in real time, allowing us to try a variety of different riff and melody ideas whilst our loop is playing. With auto disengaged, Live won’t generate any notes until we hit the generate button, meaning we can fine tune parameter settings first and then create a pattern when we’re ready.

It’s worth noting that Seed creates MIDI notes based on the key and scale set within the MIDI clip, which by default will be set to the same as the key and scale of the overall project.

Shape MIDI generator

Shape is another MIDI tool added as part of Live 12. Where Seed uses preset parameter settings to control the notes being created, Shape generates notes positioned around a custom curve shape that can be drawn or selected in the bottom left of the MIDI clip.

As with Seed, a selection of parameters allow us to set the range of the pitch, velocity and timing of the notes created. Again, we can work with auto-generate on or off in order to control whether changes are mode to the pattern in real time.

Transforming MIDI with Recombine

Alongside its generative tools, Live 12 also added a range of MIDI transformation tools. Rather than creating MIDI patterns from scratch, these allow us to generatively alter existing MIDI patterns using an assortment of parameters and processors.

Here we’re using the Recombine tool. This takes our existing MIDI notes and shuffles them, in order to create a new pattern from an existing one. Using this shuffle feature, we can rearrange the position of the notes in a pattern whilst keeping its general shape and rhythm intact.

Hardware-style sequencing with MIDI sequencers

For an alternative route to bass and lead creation, we turn to the Sequencers pack, which is included with all versions of Live, but needs to be installed via the Packs menu in the browser.

This pack contains a variety of MIDI sequencers inspired by the workflow of classic hardware. We’re using the SQ Sequencer tool, which takes its inspiration from CV sequencer tools.

This generates notes in real time based on the parameters of its steps. We can enable scale mode in order to match the key of our track and use the octave control to alter the range of notes to suit leads or basslines.

One of the most interesting aspects of the SQ Sequencer is its dice controls, which let us randomise the output of various parameters. We can randomise everything from the pitch of the steps to the rhythmic pattern itself, as well as note length and velocity.

Acid sequences with Sting

Sting is a Max for Live device that isn’t included with Live, but you can download it for free. Sting is an acid line generator, inspired by the sequences of acid house and techno, typically created by Roland’s TB-303 and SH-101.

Sting is an incredibly simple device to use. We simply set it to the correct key – or to automatically follow the key of our project – and then click the smiley face icon to generate a pattern. If we don’t like what’s created, we can hit the face again in order to create another pattern.

Various pattern settings let us control the density, direction, length and accents of the pattern. Once we have something we like, we can hit the clip button and the device will create a MIDI clip based around the current pattern.

Tom Glendinning is a session musician, sound engineer, sound designer, music producer, composer, and VJ. As an Ableton-certified trainer he helps musicians be creative with Ableton Live. Connect with him at: https://elphnt.io/

DJ and producer Benji Roth – aka Semtek and DJ Persuasion – founded the record label Don't Be Afraid in 2010 as an outlet for his own productions.

Over the years since, the label has grown to become one of the UK's most reliable underground imprints, connecting the dots between music scenes in Bristol, London and Detroit, and releasing music by the likes of rRoxymore, Mr Beatnick, Ikonika, Karen Gwyer, along with numerous tracks from Roth himself.

Last month Roth returned to his own label with a new Semtek EP, Four More Dubs. We visited Roth in his studio to quiz him on what he’s learned in the course of 15 years of creating and releasing house and techno.

For our studio session, Roth returned to his 2012 release Couer De Lyon after more than a decade, in order to explore how his production skills and approaches have changed over that time, and see if he can improve on the original version.

You can watch the session in full below. In it, Roth explores the various techniques he’s applied to the track. He also explores how he uses remote analogue gear via Access Analog, and why he likes to cut tracks to dubplates.

We’ve collected some of our favourite quick tips from the session.

Layering drum sounds

For a classic house track like this, Roth mostly sticks to staple sounds of Roland’s ’80s drum machines, but he’s not afraid to mix-and-match to find sounds that compliment one another.

“When it comes to writing classic house and techno, I'm a 909,” he says. “The reason I prefer the 909 is that I grew up playing double bass and bass guitar, so I like writing basslines, and that 808 kick doesn't leave much space for a bassline.

“I'm going to nick the cabasa off the 808 though, because I still love that. What a luxury it is to be able to pull kits in from other drum machines that you've never been able to afford.

“I'm especially a fan of the DMX just for cabasa. So I've got the DMX cabasa layered underneath the 808 cabasa and the open hat. I love the way that a cabasa or two will meld with the hats and create some texture in your high end.”

Processing a sub with hardware

The main bass sound in the track is a simple sub tone. As Roth explains, a little processing helps it cut through the mix.

“I've used a very plain sub here, possibly just a sine tone,” he explains. “The problem with that is that you're going to feel it on your monitors, and you might even feel it in your headphones, but you're not necessarily going to hear it.

“These days, I like to pre-prepare and pre-sample patches that might begin as, let's say, an Akai test tone. What I've done is I've run them through some hardware, one note at a time, and saturated them a little bit and given them some texture.

“I'll let you into a little secret about the Akai test tone. The reason I think people really liked it back in the day is because it's got a little bit of a click at the beginning, and the part of any sound that your ear is most attuned to is the attack. So that helps it to stand out.

“Luckily, my friend Borai who is based in Bristol – big up Boris – passed some of those sounds through his hardware samplers for me, one note at a time. How generous is that? So now I've got some slightly richer subs to work with.”

Separate sound design and writing

“I find that choosing a sound quickly and just getting on with it, so you can get to the writing is really helpful,” Roth says. “I like to have pure sound-making sessions where I make myself different synth sounds. That could be sampling myself playing a chord on a synth I like, then I’ll key it up on the sampler. I might bake in some reverb and delay.

“Just kind of go mad with whatever's there – software or hardware – but create sounds that you can use further down the line. The reason being that when you then come to make a tune, you're just writing, and you're not too caught up with the sound design.”

‘Head sounds’

For Roth, a key part of making a dancefloor track is adding the small, incidental elements, which he calls ‘head sounds’.

“Years and years ago, when I lived in White Chapel, I went to an open demo session that an act called Future Shock were running at Digital village,” he says. “I played them a track, and they said, 'well, it's all right, but where are the head sounds?' So I've gone, 'what are head sounds?’ They said, 'you know, just little sort of tweaks and blips and blurps'.

Roth explains that although he was applying dub effects to the drums, they recommended he add sounds designed specifically to sit in the background, and add dub effects to those too.

“One of the reasons I like that term ‘head sounds’ is it makes me think of head shops, which are the cafes in Amsterdam,” he tells us. “If you think of the kind of music they play in those places, whatever genre and or style it is, they've always got these little sounds that stick in your head.

“They're kind of hypnotic, but you might not know what they are or where they've come from. It's something you hear a lot in trip hop, and I think it's good to bring those to the dance floor as well.”

Use every part of your kit

“If you ever watch a drummer playing, you'll notice that they hit all the different skins and all the different rides and cymbals at some point. It’s very unusual that someone just sits there hitting a kick and a hi-hat and snare like that,” Roth explains.

“I think that if you want to bring a human touch to your track, pressuring yourself to try and use all the different hits in your kit, even if it's only on the fills, is a really good idea.

“In my 909 here, I've used the rims and the snares on my fills, but I've got the clap going the rest of the time, and I've got the crash on the one, which for house is a really good idea.

“If you're making more experimental music, you might say, ‘Well, why would I want a crash and ride in there?’, but you can still use the same principle and just have the ghost of a crash or a ride, or a sound that's doing a similar thing in a similar part of the frequency spectrum.”

Think about the longer arrangement

“Something that I often say to people who are writing their first project is don't get trapped by writing all your loops on a four bar cycle,” Roth says. “Don't let that become your project before you've arranged it.

“My vibe, personally, is to arrange as soon as you can. Even if you’ve only got two or three elements, you can already arrange. That's your canvas. Then you can start to think about longer events – like the strings in my track – that happen over a much longer course of time.”

Visualise the vibe

“Many years ago, I was working with an artist called Kowton [aka Joe Cowton], who was doing a remix for my label. Joe was one of the best to ever do it in that style. He'd sent me this remix, and he was saying, ‘What do you reckon?’ I was listening to it, thinking, I respect his style so much, and I respect his technique so much that I don't want to give any very specific feedback, because I don't want him to feel like I'm patronising him.

“He was a big skater, and so what I told him to do was this: I think you need to put on Hocus Pokus, which is a classic skate video we both love, turn the sound off, see if your track works with the video.

“If you've got a favourite skate video, or something where there's a cool soundtrack that you love the vibe – it could even be a music video or just the centre label of a record on a label that you really love – put it in front of you and see how your track works. I guarantee you'll be surprised by what you learn.”

Four More Dubs by Semtek is out now on Don’t Be Afraid

RECORDING WEEK 2025: Back in 2021, we spoke to four star guitarists to get some of their top tips for aspiring axe-wielders. Their advice is republished below to mark MusicRadar's Recording Week.

When it comes to recording guitar, everyone goes through a slightly different process.

Over time, we all find ways of working with comfortable 'old' setups and often end up discovering what could be considered unconventional methods. These procedural variations also factor as a major part of what creates unique music.

With this in mind, we contacted four star guitarists - John 5, Paul Gilbert, Adam Dutkiewicz and Marty Friedman - and asked them how they go about capturing their guitar performances on hard disk or tape. Their methods certainly work for them, and they might just work for you, too.

John 5

1: Go organic
On [my album Sinner], I took a more simple approach to my amp setup. At the studio where we record, they have loads of amps set up. I used a Marshall JCM 900 and a Fender Tone Master. We kind of just fire them up and go. I haven't been using a lot of effects, as I've had such a cool, organic sound - that's how I practise at home.

2: Doubling up
Everybody doubles their rhythm parts, but I always thought, 'Why don't they double their lead solos, too?'. It's really hard to do, especially with any bends and faster sections, and it does take a very long time, but I've been doing it a different way to make things a little easier.

For instance, as soon as I've just finished a song, I'll double the lead part right away, while it's still completely fresh in my mind.

3: Scouting around
Always be prepared, and know what you're doing. It's like when you're in school; you'll want to study before you do a test, if you're going to do any good in it that is. Well, it's the same thing with anything in life, and recording guitar is definitely included.

4: Limiting factors
Go straight (direct input), or at least try to limit yourself to only one or two pedals or effects, otherwise you'll only mask your true sound and take something away.

www.john-5.com

Paul Gilbert

1: Mic 'n' easy
Electric guitar is the easiest thing to record. It's not like drums or vocals. Distorted guitar is especially easy, as it's already compressed. Use a dynamic mic, don't do anything fancy with it - just point it straight at the amp or angle it slightly and it'll sound great.

2: Tuning and timing
Recording comes down to two things: tuning and timing. Lots of unsigned bands I hear overlook timing. People think AC/DC, Green Day and the Ramones aren't great players because they don't play fast, but they have their timing nailed and their records are really tight and sound great as a result.

3: Balancing act
Editing parts is okay, but you need to find a balance and know where to draw the line, otherwise you can waste hours. I used to find that, after spending so much time editing, I lost sight of the original feeling that I was trying to create. You should always try to get the best possible performance in the first place.

4: Under-doing it
You can get lots of distortion from any amp or plug-in - it's not like 1975 when Marshalls just had a volume control. The problem is that jacked up guitars with loads of distortion will sound like mush. Try to get the song sounding good without loads of distortion and overdubs.

www.paulgilbert.com

Adam Dutkiewicz

1: SM57's rule
You either call it 'knowing what works' or blame it on laziness, but I always just throw a '57 in front of the cab. I suppose it's a 'stick to your guns' mentality. I default to that every single time.

2: Back off
Mic placement is pretty crucial. You can get a million different EQ responses depending on where you throw the mic in front of the cab. I personally have the best luck - or at least I think so - when I back the mic off a little bit.

I know a lot of engineers throw it right on the grill to get the bass boost, the proximity effect and all that garbage, but I find that if I back it up about six inches, I get a more balanced EQ curve.

3: Tight fit
Half the battle is playing the part well. Remember the old phrase, 'You can't polish a turd'? If it sucks from the beginning, it'll suck at the end. There's no way around a bad sounding performance.

4: A little bit at a time
My brain is not intelligent enough to perform a song well in one pass, so a little bit at a time works better. That way, the player can concentrate on what's going on at that specific moment. It's too overwhelming to play a whole song good.

https://killswitchengage.com/

Marty Friedman

1: Prepare…
Prepare and over-prepare, and then prepare some more. When you're done, practise everything over again, just once more. Then, by the time you get in the studio everything will just flow out of you.

2: …Or not
Despite what I just said, sometimes I feel that being totally unprepared makes for a really fun, fresh experience. If you have an unlimited amount of time, you can be as unprepared as you want and simply hope to get lucky.

3: Refined sugar
I'll have one go at an entire solo and even when there are some parts I like and others I don't, I won't punch in and get caught up in little details. I'll just remember the parts I like and then replay the whole thing again. I'll repeat that process, so the final solo ends up being one take. Engineers like doing it that way, too.

4: No one approach
Steve Vai and John Petrucci of Dream Theater played on my 2006 album Loudspeaker. John came into the studio, where I walked him through the entire track, then got him to play all the impossible stuff I couldn't imagine doing myself. Steve had no direction; I sent him the song and he sent a bunch of tracks back, which I hacked up to my liking.

www.martyfriedman.com

RECORDING WEEK 2025: These days, producers and artists have access to music-making tools of such high quality and versatility that creating a decent-sounding bassline might seem like the easiest thing in the world.

But the problem, of course, is that everyone has access to these tools. As you marvel at how simple it is to fire up a bass preset on a synth, program a one-note pattern and loop it across your track, bear in mind that all of your contemporaries will have the option to take this path of least resistance, too - which means the ones who’ll stand out are those who work a bit harder.

To put it another way, we’d argue that you should give your bass parts the attention they deserve, as otherwise they run the risk of sounding at best generic, and at worst pretty lousy.

Hence we present the following 37 tips: these bite-sized chunks of advice are designed to make you think again about the way you approach composing, recording, programming, arranging and mixing your basslines. With any luck, they’ll help you to do at least some of these things better.

There’s no magic formula here - hitting the low-end sweet spot can be as much about trial and error as it is about flashes of inspiration - but what we can say is that a great bassline is the sum of many parts. You might have come up with a floor-shaking sound, but what good is that if you haven’t programmed a decent groove? Alternatively, your ear for a catchy bassline might be well trained, but if you can’t make yours sit properly in the mix, its earworm properties will be lost.

Winning the bass race, then, isn’t going to be easy – but with as your personal trainer, you’re off to a flying start.

Designing bass sounds

1. Think different 

If you’re struggling to get the kind of bass sound that you’re looking for, why not think out of the box and try something completely different? It’s not unheard of for producers to use pitched-down woodwind instruments to create basslines – try a bass clarinet, for example. You might need to apply a bit of processing, but this can create a nice, buzzy bass tone.

2. Easy on the reverb

If you think your bassline needs a little more ambience or depth, it can be tempting to throw a reverb over it. Our advice is to tread carefully in this area, as doing that can often do more harm than good. Other effects – panning or chorus, for example – are usually more suitable, and you can get good results from layering up bass sounds, too.

3. Strip it back…

Most synthesizers these days come pre-loaded with a slew of high-quality bass presets, many of which sound big enough to cause a minor earthquake. Sometimes, though, less is definitely more, and you might find that you’re better off pulling out some basic synthesis techniques and going to work on simple oscillator waveforms.

4. Filter it out

That said, even if you’re drawn to musical styles that regularly feature sine-wave basslines, it can be better to use a core sound that’s richer in frequency content and then use a low-pass filter to remove unwanted harmonics. Doing this means that you can automate filter changes as your track progresses. A sine wave contains no harmonics, so there’s nothing to filter.

5. Open the envelope

Once you’ve got your low-pass filter set to the right position, try using an envelope routed to the cutoff to open the sound very briefly as each new note is played. This provides extra articulation so that the listener will really be able to hear where each bass note plays. The best settings to use when doing this are short attack and decay times, and no sustain.

6. Attack, attack, attack!

Getting your amplifier envelope settings right is crucial. If you’re emulating an electric bass, for example, go for a fast attack, long decay, no sustain and a fairly fast release. You can make your own choices when you’re creating a synthetic tone but, as a general rule, nearly all basslines need fast attack times. A bass that sounds like it’s coming in late is not good.

7. Get clean

Sampling basslines from existing tracks is common practice, but it can sometimes be difficult to capture a clean groove that doesn’t feature any other parts. Try using a low-pass filter to get rid of the high-frequency sounds, leaving you with just the bottom end.

8. The art of noise

See that noise generator on your favourite bass synth? As the name suggests, it’s not there to keep quiet; try mixing a bit of white or pink noise in with your main bass sound to keep the interest levels up. This doesn’t have to be there all the way through, but it can make a nice addition during an alternate section such as a bridge or breakdown.

9. Make it in mono

It’s usually better to create your basslines in mono rather than stereo, so that they come from a fixed, narrow source. It might sound dull, but panning centrally is generally advisable, too. If you are working with a stereo synth preset, look carefully at the internal effects it uses and turn off the ones that add stereo width – unless width is something you’re actively going for.

10. Automated pump

Do you want to get a pumping bassline without the bother of sidechaining a compressor? The quick and dirty way to do it is to automate the gain of your bassline so that it rises and falls rhythmically, though it should be said that this technique might not give such smooth results as doing the job properly would.

11. Sound design time

Perfecting your bass sound can be an engrossing process, but it can become frustrating if you’re trying to do it in a session when you’re also working on the composition of your bassline. Make sure you set aside dedicated time to get into the finer details of your bass tone.

Programming bass

12. Play it not so straight

Quantise can be very useful for tidying up recordings or locking a bassline to the grid, but don’t be a slave to it. Some basslines will benefit from being left ‘as is’, while others are better ‘creatively quantised’ using the groove template features of your DAW. Don’t forget the good old swing function, either.

13. Slide away

For creating smooth basslines, with the pitch of one note sliding nicely into another, your synth’s portamento control (also known as glide) is your friend. Engaging it should give you the desired effect, and it can work wonders on electric, acoustic and synthesised basslines. Don’t feel like you have to use it throughout the whole track, though – it can have more impact when dropped in at opportune moments using automation.

14. Perfect pair

The relationship between your kick and bass will be instrumental in the success of most tracks, and it’s something that is really best nailed down early on in the production process of your song. This is because the combination is most likely going to be your track’s ‘engine room’, as it were, and getting it right will give you a solid base on which to build the rest of your tune.

15. Hands on

There are loads of products on the market that enable you to add typical bass-playing ‘extra’ sounds (picks, string squeaks, scratches, etc) to your parts, but if you don’t have a suitable plugin, there’s nothing to stop you doing it manually instead. Layering up samples of such sounds on certain notes can be great for accenting them – simply place them on a new audio track and adjust the volume to taste.

16. Learn from the best

They say that imitation is the sincerest form of flattery, so if you have a bassline in mind that you want the one you’re working on to sound like, why not try emulating it first to get a feel for its groove and musical structure? Once you’ve done this, you might find that you can come up with the bassline you’ve been looking for just by moving a few notes around in your MIDI editor. Outright plagiarism is probably best avoided, though.

17. Double up

There may be occasions when you want your bassline to mimic another part of your arrangement exactly in order to emphasise it. In these cases, there’s no need to record the bassline from scratch: simply copy the MIDI data from the existing track to your bassline’s. You may need to drop it down an octave or two, but doing this should save you some time.

18. Pitch-perfect

Deep basslines are crucial to some genres, but it can be hard to hear if the one you’re creating actually fits with the rest of your track when it’s pitched down so low. If you’re wrestling with this problem, try pitching your bass patch up a few octaves to make sure you’re in the right key and on the right track.

19. Just dropped in

Similarly, you can actually go so far as to replace a few MIDI-triggered bass notes in a phrase with real ones from a recorded bassline. Chop them up and drop them in where they work best, but don’t forget to use short fades at the ends of these notes to avoid clicks.

20. Let’s gate it on

Don’t underestimate the usefulness of gating. Try applying a gate to an unruly bassline and feeding your kick drum channel into its sidechain input. This will ensure that your bass notes and kicks trigger simultaneously for a punchy, tight, clubby sound.

21. Hear the harmonies

Your bassline will often follow the root notes of the chords being played by other instruments, but experimenting with other notes can have real benefits. The whole flavour of a song can be transformed if you change the bass notes to others that harmonise with your chords.

Arranging and mixing bass

22. Delayed gratification

Many of the best basslines are practically hooks in themselves, but don’t feel like you always have to show off the Crown Jewels immediately. Sure, a bassline can be a great introduction to a track, but there’s a lot to be said for holding it back for a few bars to build anticipation, too. Also, try starting with a simplified version of the bassline and building up to the full groove gradually.

23. Filter fiddling

If you want to introduce a big bassline as part of a drop, one well-worn trick is to use a high-pass filter during the eight bars (for example) leading up to it. If you automate this so that the lower frequencies of the sound are introduced gradually, you’ll build anticipation and create a bigger impact when the full sound kicks in.

24. Keep changin’

Even if you think you’ve got the best bassline in the world, try to resist the temptation just to loop it across your entire track and leave it as it is. Subtle changes to note timings at certain points can have a greater impact than you might expect, and little flourishes at the end of every four- or eight-bar section (or just before a chorus, for example) can help to keep things interesting for the listener.

25. Sort out your speakers…

Everyone knows that a decent monitoring setup is crucial for fruitful mixing, but this is particularly true for basslines. You won’t really be able to hear what’s going on with your low-end frequencies if you don’t have a good set of speakers (or, at a push, some high-quality production headphones), so investing in some is highly recommended.

26.  …And your sub

One other option is to add a sub-woofer to your setup. This will enable you to get a much better feel for how your bassline’s sub frequencies are working. If you can’t stretch to one of these, a spectrum analyser can give you a visual guide to what’s going on down there.

27. The clash

Although the bass guitar and kick drum can be best friends, their frequencies sometimes clash. If you’re having this problem, try notching out some of the kick frequencies from the bass part. A spectrum analyser can help you to find these (try starting your search around 90Hz). Once you’ve located them, use a narrow Q to pull a few dB from the bass.

28. Get on the buss

As we’ve mentioned elsewhere, it’s common practice to create a bassline using multiple layers. When the time comes to start mixing, it can be very useful to route these layers to a single buss so that you can apply processing to them together while also retaining the ability to tweak each layer individually.

29. Fill it up…

Is your bassline lacking fullness? Try sweeping a bell-shaped EQ boost between around 500Hz down to 150Hz. A broad boost of just a few dB can often be enough to provide the low-mid glue that’s required.

30.  …Or empty it out

You might also find that your mix is sounding too full in this same range. In this situation, it might be the case that the frequencies of other instruments – pianos, guitars and synth pads, for example – are cramping your bass’s style, so each one will require individual attention in order to strip things back a bit.

Recording bass

31. Tune up

It sounds obvious, but you’d be amazed at the number of recordings we hear that feature an out-of-tune bassline. There really is no excuse for it, particularly when your DAW will either have a built-in tuner or be capable of hosting a plugin that does the job.

32. Buzz off

That buzz you can hear on your bass recording might initially sound like it adds character, but it’ll quickly become annoying. If you’re getting noise in your signal when you record, try moving your bass player around, as the likes of monitors, lights, mobile phones and ground loops can all cause it.

33. Same but different

It’s easy to assume that all bass guitars are born equal, but you’ll get a different type of sound depending on the instrument that you’re recording. Fender Precision and Rickenbacker 4000 series basses are favoured by rock acts, for example, while the likes of the Fender Jazz and Music Man StingRay tend to be used by pop and funk artists. So, if your player has several different guitars, make sure he or she uses the right one for the job.

34. Which way to play?

Similarly, there are several bass-playing techniques, each of which will give you a decidedly different kind of sound. Fingered bass is arguably the most versatile, providing a full, rounded tone, while picked bass has plenty of attack and cuts through the mix. And then, of course, there’s slap bass, which in the wrong hands can turn out very badly indeed.

35. Best of both worlds

People frequently ask if it’s better to record a clean DI’d output or an amplified one, but the truth is that it’s often better to use both signals. If possible, you should keep the two separate at the recording stage so that they can be mixed together later. An alternative option is to record just the DI signal, then feed it into an amp simulator that enables you to blend it with the processed sound, as some of them do.

36. Record everything

If your bass player is having a few practice runs prior to recording their part for real, why not try surreptitiously hitting the red button and getting these down as well? Some musicians can put too much pressure on themselves when they’re ‘going for one’, and you might find that the loose dummy run ends up sounding better than the ‘real’ recordings.

37. Mess with your mics

There are various accepted ways to record a bass amp, but if you want to get creative, try sticking a mic round the back of the amp as well. This can be used to capture a phase-inverted version of the front mic, and can lead to some very interesting sounds.

For many electronic musicians, music theory isn’t something that comes naturally. It’s perfectly possible to know your DAW, sequencer or synth inside and out, but have only a basic grasp of concepts such as chord shapes, scales and harmonic relationships.

If you’re a producer that fits this description, it can sometimes be hard to create chord progressions and melodic parts that reach the same heights as your sound design skills. You may know how to input a series of simple chords into a piano roll, but what happens when you want to try something new or more adventurous? How do you venture beyond your tried and tested combinations of chords?

Ableton Live 12 offers a multitude of tools that can be used to work around gaps in your theory knowledge. In our latest tutorial video, Certified Trainer Tom Glendinning explains a range of methods that Live 12’s packs, devices and generative tools can help users explore fresh chord shapes and combinations.

Expressive Chords

Expressive Chords is a pack that comes free with Live 12 Intro, Standard and Suite. It’s not part of the factory library, so to access it you’ll first need to head to the Packs menu in the browser and download and install it.

The pack works by providing a range of preset chord banks. Each chord in these banks can be triggered using a different note play via a MIDI keyboard or any other note input. Expressive Chords comes with a variety of presets based around styles and genres, with chord banks configured to suit a specific sound.

Below the main chord grid is a range of controls altering how the chords are played back. Tilt can be used to emphasise the higher or lower notes in a chord – creating a brighter or darker sound. Invert will create an inversion of the basic chord shape, moving lower notes to the top of the chord.

Strum spreads out the timing of each note in the chord, creating a guitar-like strumming effect. An articulation editor lets users go deeper into how a strum is played by adjusting the spacing of each individual note.

The ‘expressive’ part of Expressive Chords comes in the fact that it is MPE-ready, allowing users to apply modulation and articulation to individual notes in each chord, via a compatible controller or MIDI clip automation.

One thing worth noting is that the chord progressions created by Expressive Chords don’t track the global key settings in Live.

Creating inversions

If you have a simple progression recorded within a MIDI clip, creating inversions on some of the chords is a quick and easy way to add variation and interest.

An inversion is created when we take a note from a basic chord shape and move it up or down an octave, essentially reordering the notes in the chord without changing the actual notes used. That’s easily done in Live’s MIDI editor by selecting a note, holding shift and pressing up or down on the keyboard.

In the video example, a simple one-bar loop is turned into a more interesting two-bar progression by creating inversions in the second bar.

Generative MIDI chords

Ableton Live’s MIDI effect devices include a variety of tools that not only allow us to randomise the notes of incoming MIDI, but also to control that randomisation. By combining these we can generate and capture randomised chords around a set scale.

In the video, Tom does this by combining Live’s Chord, Random and Scale devices, placed in a chain in that order.

Now, whenever a single note is played, the Chord device turns that into a chord based on the intervals set on the device. By default, those intervals and the resulting chord shape will remain the same. no matter what root note is played.

By placing the Random MIDI device after Chord, we can randomise the notes coming out of that device. There’s a percentage control here that adjusts how likely it is that an incoming note will be randomised. At 100% every single note coming from Chord will be randomised to a new pitch. At less than 100%, some notes will remain from the pre-defined chord, while others will be randomly changed to a new pitch.

Using these two devices alone, the results will sound random and messy, not adhering to any specific scale or pattern. By placing the Scale device at the end of the chain, we can make sure all the incoming MIDI notes are mapped to our chosen scale/key.

Now, by playing single notes, we can randomly generate a chord progression. In the video, Tom captures several bars of randomised chords and then edits together a loop of his favourites.

Tom Glendinning is a session musician, sound engineer, sound designer, music producer, composer, and VJ. As an Ableton-certified trainer he helps musicians be creative with Ableton Live. Connect with him at: https://elphnt.io/

Spotify Wrapped is just six weeks away. I know. Where has this year gone?

For punters, the platform’s Wrapped week is a mirror showing who or what exactly you’ve spent your time listening to during the last year, but for artists and musicians it’s a chance to engage with your fanbase and keep them informed of what you’ve got coming up in 2026.

Why? Well, according to Spotify, 245 million listeners engage with Wrapped and there are some 435 million shares of Wrapped content. Those figures alone justify the time and effort.

So you could – if you so wish – upload a personal clip for fans. All artists have the opportunity to do so, though keep in mind that the deadline is November 14.

The clip must be 30 seconds or less. That’s not much time to say thank you to your listeners, and perhaps tease what you’ve got coming up next year. If there’s nothing in the immediate pipeline, you could share a milestone from the year just gone, a particularly memorable fan encounter or a moment that you’re especially proud of.

Once you’ve recorded your clip, head over to the Video & Visuals tab on Spotify For Artists and upload there. The video will go live when Wrapped launches towards the end of November and will stay there until the campaign ends a week later.

As well as the clip, it’s important to keep your artist page updated at this time. So make sure your page contains all your upcoming gigs as well as details of ticketing partners.

If you do have a release coming up it might be an idea to create a Countdown Page to your next album or EP, making it easy for fans to pre-save via your artist page. When the release finally does drop , those pre-savers will be notified and your music will go straight into their libraries.

Then there’s your artist profile. The run-up to Wrapped is the perfect time to make it sparkle. You could use Artist Pick to pin your favourite music video of the last year, or add your own best of 2025 track list. Also make sure that fans looking at your profile can easily locate your merch and purchase during the peak Wrapped week. It might even be time to refresh your photo and put a new look front and centre.

Yes, there’s a lot to do. But remember those stats – nearly every Spotify user will engage with Wrapped in some way. It’s a massive opportunity for artists.

RECORDING WEEK 2025: These days, it’s perfectly possible to create the majority of your sounds inside your computer, but if you never bother to capture any real-world audio, you and your music will be missing out.

With this in mind, MusicRadar has amassed 25 timeworn tips for anyone who has an interest in live or studio recording.

1. Get mic'd

Buy yourself at least one decent condenser and one decent dynamic mic as soon as you can. As long as you don’t juggle with them, they’ll last you for years and open up a whole world of fun. Music production is largely about recording stuff - don’t lose sight of that.

2. Go live

Don’t dismiss live recording. With so many samples readily available, it’s very easy to overlook it, but you must try it a few times, as it teaches incredibly useful skills and highlights a lot of principles computer musicians often ignore, such as phasing issues and other mixdown-stage problems. Oh, and it’s fun!

3. Two mics, one guitar

When recording guitars, try using two mics placed centrally but pointing at opposite ends of the instrument. They’ll produce tonally different sounds that can then be panned later for a great stereo effect.

4. Put down a marker

If your musician is swinging their instrument around too much it will affect the sonic quality, no matter how much compression is applied, so it’s sometimes worth hanging a marker or dummy mic and asking the player to keep their instrument level with it.

5. Keep your distance

Live recording and studio recording require different approaches. Live recording is all about close-miking to avoid noise bleed, but this almost always sounds worse than a mic placed a comfortable distance away, so don’t be tempted to close mic in the studio…

6. Be prepared for inspiration

Create a recording template project in your DAW that is pre-set to your typical recording preferences. With that done, whether you’re a musician, DJ or just an engineer for others, you’ll always be poised to start recording any flashes of inspiration.

7. Balance before buttons

When recording a group of musicians with one mic (a brass section, say), the key is to create the right dynamic balance before you hit the record button. Try placing your players about three metres from the mic and pointing it down at them slightly, so as to capture reflections from the floor.

8. Put the listener in the room

Essential to the successful recording of acoustic instruments is creating the illusion of being in the room. Your ultimate goal is to make it sound like it would to your ears in real life. That is what makes it such a challenge - the whole art is in making up for the fact that no microphone captures sound like our ears do.

9. Move the mic

If you find mechanical noise is clouding a recording (saxophone keys clicking, for example), try placing the mic behind the player or to the side, so that it's partially shielded from the offending cacophony.

10. Keep acoustic mics at 35-45cm

The ideal distance for an acoustic guitar mic is around 35-45cm away from the soundhole. Place it any closer and your sound will be overly boxy - much further away and you´ll pick up too much of the room´s ambience.

11. Try new styles

Get your musician to try playing in a number of styles (eg, picked and fingered for a guitarist). You never know which might sound better, and you then have layering options for mixdown and different versions for your arrangement stage.

12. 'Verb the vocalists

If your vocalist is struggling to hit the notes, try washing their monitoring signal with a lot of long reverb. They’ll then be able to use the reverb on their voice to better detect their own pitch, making it much easier for them to make adjustments.

13. Double mic pianos

A handy tip for school music teachers: the best way to record an upright piano is with two condenser mics behind, placed left and right and pointing slightly down, with the top and back of the piano removed. If you can’t remove the back, open the top and place the mics left and right, about two feet above.

14. Amp your electric piano

If you’re using a classic electric piano riff, try recording it using an amp or amp modelling plug-in for that authentic live stage sound.

15. Bend time

Struggling with the performance of a riff? Thanks to timestretching, you can play it slower and more accurately, then ramp it up to the right speed later. It’s not cheating if nobody knows and it sounds great!

16. Listen for natural reverb

Does your building have an interesting-sounding hallway or other acoustic space? A really bright-sounding bathroom, perhaps? Put a microphone in there, play your track back and record the room’s natural reverb, ready to mix in.

17. DIY vocal booth

For decent vocal recordings, create a small area packed with duvets and blankets - a wardrobe is ideal. Of course, you need to ensure that you have enough light to read the lyrics and enough space to be able to stand a decent distance from the microphone.

18. Use closed-back headphones

It sounds incredibly obvious but when you’re recording an instrument or voice, make sure that the backing track only plays through well-isolated closed-back earphones with minimal noise-bleed, so as to record a clean signal.

19. Add in ad-libs

Always get loads of ad-libs from your vocalist, and make sure they try singing sections of verses and choruses in separate takes, holding notes so that the tail of one vocal section can be laid out to overlap the start of the next. This can create a really energetic effect.

20. Mic from above

The best placement for a vocal mic is actually slightly above your vocalist, as it forces them to tilt their head back slightly, thus opening up their throat for clear and strong voice projection.

21. Keep them off the booze

Whatever you do, don’t let your vocalist get drunk or worse. There might be a few rare musicians out there who do a better job after they’ve had a few beers, but they are a very small minority, and this is especially true for vocalists… whatever they might tell you!

22. Look out for latency

Recording guitars can be a tricky business because of latency issues in some setups. If you like to monitor with effects, but want to record a clean signal, then you need to use a unit that enables you to output the dry signal alongside the effected version.

An alternative solution is to get hold of a small analogue mixer/splitter and send the signal to your effects unit and the computer simultaneously.

23. Become a vocal coach

If your vocalist just isn’t hitting the right notes, try doing a little DIY vocal coaching. For example, if they’re singing flat, ask them to sing a little sharp. And if they can’t do that, try temporarily tuning the backing track up a few cents (singers tend to be around 10-20 cents sharp or flat) to force them to compensate.

24. Record everything

We’ve already told you to make yourself a recording template, but that’s no good if you don’t use it. You should get into the habit of leaving something recording every time you play an audio instrument, mix, sing or otherwise jam. Then just delete it if you haven’t done anything worth keeping.

25. Rattle and hum

If you’re recording with sensitive microphones, make sure your players aren’t wearing anything that might rattle or bash against their instruments. There’s nothing more annoying than the sound of a watch, button or jangling chain ruining that one perfect take and remaining undiscovered until after the musicians have gone home!

RECORDING WEEK 2025: Most people with an interest in music production or recording probably have at least a loose idea of what compression does. Put simply, it ‘squeezes’ incoming audio by reducing the dynamic peaks in an audio signal over a certain threshold.

This reduces the dynamic range of the audio – ie the difference between the loudest parts and the quietest parts – and as a result can make the overall level sound louder, since you can raise the volume of a track without worrying about clipping.

When done wrong, however, over-enthusiastic compression can leave music sounding flat and boring.

For newer producers – and, let’s be honest, also us more experienced ones at times – compression can be a bit of a confusing technique. Tutorials on the subject often tell people to ‘trust your ears’ or apply compression ‘to taste’. But that’s often not helpful for those without the requisite experience to know where to start.

To that end, treat this article as your ‘cheat sheet’. In the graphics on this page we’re outlining the basic settings we’d use for a variety of different compression applications.

These do, however, come with a fairly sizable caveat: don’t take these examples as hard and fast rules. More accurately, think of them like presets - they’re starting points you may want to work from.

Compression and other dynamics processes should be tailored to the audio they’re being fed, so feel free to break away from what we’ve outlined here if it sounds right.

Ultimately (uh, oh… here it comes) trust your ears, and if a different setting sounds better – or even if it sounds better with no compression at all – go with what you think is working best.

Compression controls explained

While different compressors often work slightly differently, and controls on offer can vary, there are certain parameters that are common across most effects.

Threshold is used to set the level where the effect kicks in. Any audio that rises in volume above this threshold will be compressed (ie reduced).

The amount that the audio is compressed is determined by the Ratio. A ratio of 2:1, for example, would mean that any audio above the threshold is reduced in volume by half. Exceed your threshold by 2dB, and the signal would be reduced by 1dB; exceed it by 6dB, and the signal would be reduced by 3dB.

Higher ratios mean more aggressive compression, all the way up to ‘infinite’, which results in hard limiting, where any audio exceeding the threshold is cut off completely.

Some compressors also allow something called ‘upward’ compression. This is essentially the opposite of what's described above, where volume is boosted when the level falls below the threshold.

Attack and Release control how abruptly compression is applied and cut off. Attack controls how quickly the reduction in level is applied once the audio breaches the threshold. An attack of 0 would mean the compressor kicks in immediately, while a longer attack will cause the compression to fade in.

Release controls how quickly gain reduction ends once the audio drops back below the threshold. A release time of 0 would mean that the compression stops immediately, longer release times mean that the compression level is reduced more slowly.

Many compressors also feature a Sidechain input. This allows for the compressor to be triggered by an external signal. The most common use of this is using a kick drum to duck the level of a bassline, to avoid the two elements clashing when they play at the same time.

What are the different types of compressor?

Not all compressors are born equal. But why do they sound different at all? And how do you know when to use which ‘type’?

Dating back to vintage hardware devices, there are a few different types of detector/gain control circuit, these are often emulated in modern plugins: VCA (Voltage-Controlled Amp), FET (Field Effect Transistor), variable mu (based on valve plate voltages) and opto (lamp plus photocell).

VCA designs can be very flexible but often the flexibility makes them harder to use. FET designs are responsive but distort sooner, although that distortion is very musical. Opto designs have a softer, more musical attack and tend to let fast transients slip through.

Finally, variable mu can be very fast in attack, however threshold, ratio and knee shape tend to all be very closely interrelated. Some units also have an automatic release option, where the release time changes to suit the incoming signal.

In the plugin world, you’ll also come across multiband compression. This breaks the incoming signal down into different frequency bands and applies compression to these separately, with results that are more like a mix of compression and EQ.

Does every sound in a mix need compression?

The short answer to this is no. If something sounds good without compression applied, then don't think you need to slap a compressor on your track just to because it's what the pros do.

Bear in mind too, that other effects can have a similar result. Subtle saturation, for example, is often just as good as compression when it comes to taming peaks, and can add pleasing harmonics too.

RECORDING WEEK 2025: While many DAW-based musicians in 2025 tend to deal with electronic sounds and loops, there is nothing like the addition of a real life human element to really bring a project to life. Thankfully, plugging in a microphone and recording audio is now simpler than ever, but the way you approach this could make or break your project, and the chances are this begins with a microphone.

If we put our technical heads on for a moment, a microphone is a form of transducer; this technical term describes a device that converts one form of energy to another. In essence, the sound which we hear is a form of wave, manifesting as pressure variations in the air.

The ‘air’ element is important, because if you tried playing a musical instrument in a vacuum, you wouldn't be able to hear it. (We’ll gloss over the lack of oxygen, but you get the point!)

Microphones contain a thin and flexible material called a diaphragm. When the pressure variations in the air hit the diaphragm, these vibrations are converted into an electrical signal, hence the conversion from acoustic energy to electrical energy.

Once converted to electrical energy, we capture that electrical signal with recording equipment, process it to shape the sound, and eventually convert it back into acoustic energy through speakers or headphones.

Pure pop
If you have ever sung or spoken into a microphone, you will probably be well-versed with the concept of ‘popping’.

This is created when a performer uses words that contain plosives, such as ‘Ps’ or ‘Bs’. A sharp blast of acoustic energy, masquerading as wind, hits the diaphragm with great force, much like a tidal wave hitting a shoreline. The resulting sound is described as a ‘pop’, and is normally an undesirable element in the recording process. However, it's also an important component for anyone who beatboxes, as it can be relied upon to mimic the sound of a kick drum.

It's also noticeable that the closer you get to a microphone, the more you tend to exaggerate low frequency content. Not so good for singer/songwriters, but excellent if you're beatboxing, unless a singer is trying to create a very intimate sound where the closeness can be incredibly endearing.

If unwanted popping is a problem for you, the easiest fix is to purchase a pop shield. This simple mesh filter is clipped in front of the microphone, allowing sound to pass through while dispersing sudden bursts of air.

Going polar
All microphones employ what is known as a polar pattern. This defines how much signal will be picked up by the microphone, around the 360° of the diaphragm.

The most common polar pattern is described as cardiod. If you imagine the shape of a heart, cast as an image across the front of the diaphragm, it would indicate that the bottom of the heart is where the microphone picks up the most signal. Around the back of the diaphragm, little-to-no signal will be captured.

An omnidirectional microphone will pick up equal sound all the way around the diaphragm, whereas a figure-of-8 polar pattern embraces the signal on either side, just like the pattern of a number 8.

Types
If you are choosing a microphone for the first time, or working to a budget, you probably want some form of microphone which can be used for all things. The good news is, most decent microphones will do a fairly good job across the board, though you should be mindful of their limitations and handle them carefully to avoid damage.

Broadly speaking, microphones fall into three main categories; Dynamic, condenser and ribbon.

Dynamic microphones are by far the most common, and are undoubtedly the most robust microphones available. If you perform live, the chances are you will use a dynamic microphone. It's safe to say that 99% of the time, live performance will involve the use of dynamic mics.

There are some very tried-and-tested dynamic designs which have been around for many years.

The ubiquitous Shure SM57/58 are highly recognisable. The SM57 is very common as a drum and guitar mic, while the SM58 is an eternal classic for vocals and frontline instrumentation.

It lends itself particularly to vocals, as it is equipped with a rugged pop shield, which does a reasonable job of preventing pops in a live situation. In fact, the 57 and 58 are so versatile, they will do a fantastic job in most recording environments. You could even place one in a bass drum, and it wouldn't complain!

One thing that you will find, if using dynamic microphones for recording, is that you may have to boost the input signal considerably. Dynamic microphones are passive in design and do not require additional power to operate. Therefore, they don't lend themselves to quieter recording scenarios, where background noise/hiss may become invasive.

This is where the condenser microphone slides into view. Condensers are normally the microphone of choice when it comes to recording.

While they operate in much the same way as dynamic microphones, they also employ a capacitor (aka condenser) to convert the sound into an electrical signal. This means that the microphone requires a form of powering, which is supplied from audio interfaces and mixing desks in the form of Phantom Power.

This electricity supply conveniently runs down the microphone cable at 48v, which is just enough power for the mic to do its thing. Some more elaborate or expensive condenser microphones may have their own external power supply, but phantom power is by far the most common option.

Because condenser microphones are so much more sensitive, they are perfect for recording quieter content, or could be used to record ambiently, where they are placed a considerable distance from the instrument source. They are incredibly versatile, but do be aware that placing condensers too close to exceptionally loud instrumentation, such as very loud guitar amplifiers, could result in damage to the microphone. It's rare, but it can happen!

Condensers also fall into two sub-categories; large diaphragm and small diaphragm condensers. Large diaphragm condensers do tend to look fairly substantial in their size. Small condensers, on the other hand, are often smaller and longer, sometimes described as pencil microphones.

They're not normally as small as pencils, but the visual association is understandable. They can also be a little more convenient in certain mic settings, where space around an instrument is enclosed or challenging.

Ribbon microphones are a category of microphone that falls between dynamic and condenser. As the name suggests, they employ a thin ribbon (often made from aluminium) which operates as both diaphragm and sound transducer.

Characteristically, they often provide a beautiful warm and rounded sound, which many describe as having a vintage character. They also tend to operate in a figure-of-8 polar pattern, meaning that they capture a signal from both sides of the microphone. This often means that the resulting sound is very open and natural.

Note that Ribbon mics can be damaged if exposed to phantom power, so you have to be careful when using alongside condenser microphones. They also emit a very low signal output, which will need amplifying considerably. This can be done using the gain on a microphone input, although there are devices that will help lift the signal externally too. These often require phantom power, eliminating the possibility of accidental damage to the microphone.

Ribbon microphones tend to exhibit a reduced frequency response, particularly in the top end. This means that they sound fantastic as overheads on drum kits, or on high strings as they don't exaggerate the brighter elements. They are also particularly good for recording brass and saxophones, or even pianos.

A few faves…
With so many mics available, it's difficult to know where to begin, so here are a few classic models to consider, all with a proven track record.

Shure SM57 and SM58 (£91/£115) - The eternal classic dynamic microphones, which are perfect for live, and useful for recording.

Shure SM7(B) (£459) - Another classic dynamic that was used by Michael Jackson, for many of his studio albums. It's also a popular podcasting choice

Aston Origin (£195) - A condenser mic, the Origin has become a contemporary classic. With a fixed cardioid polar pattern, it's excellent for vocal and instrumental recording.

Neumann TLM 103 (£749) - One of the more affordable microphone options from Neumann, it also employs a fixed cardioid polar pattern, using the same capsule as the infamous U87.

AKG-C414 (£859) - The humble 414 is another eternal classic, popular for recording acoustic guitar, acoustic piano, drum overheads and woodwinds/sax. It's become particularly popular for recording felt piano.

Neumann U87(Ai) (£2550) - One of the most iconic microphones in existence, the U87 has garnered a reputation as a formidable vocal mic over the last 50 years. It also sounds superb on just about everything else, from double bass up to piccolo! (Some contemporary engineers suggest that the U87 is a little bland by todays standards, with more colourful options available at this price point.)

Coles 4038 (£1099) - The Coles is about as British as ribbon microphones can get, having been a BBC classic for many years. It'll sound superb in so many settings, but works particularly well on brass and high strings. Rich and warm!

Royer R-121 (£1399) - Regarded by many as an underrated contemporary classic, the Royer is a beautiful example of a ribbon microphone, ideal for many string and brass based recording environments.

AEA R88A (£2899) - This mic is not only ribbon based, but stereo too. Ideal for recording small ensembles or to place over a piano for an amazingly colourful sound.

There are numerous reasons why you might want to stress test your laptop for music production. You might want to understand the limits of how many tracks or plugins you can run in a session, how long you can push the boundaries of your budget laptop before it overheats, or if you’re on a Windows laptop, test a recent upgrade.

As the most important component for music makers, your CPU has a large effect on a multitude of aspects of music production. Every plugin you use takes a little slice of your CPU power, and the heft of your CPU also affects the ability to run lower buffer sizes in your DAW, enabling lower latency performance, which is crucial in the recording phase.

One of my first jobs was working in a computer repair shop and while the majority of my time was spent dealing with software issues, quite often I’d have customers coming in to upgrade or overclock their machines, seeking better performance. It’s all well and good buying a new laptop, upgrading your RAM, or adding an external hard drive to enhance performance, but how do you actually know if these upgrades or new purchases are having a marked difference?

This is where testing comes into play. It’s a relatively simple process to undertake, and can be done with stock plugins or using stress testing software available for free, so all you need to test your laptop is a little bit of time. One thing to note is that if you’re performing this testing to see what impact an upgrade will have, make sure to do some baseline testing before you install your new RAM or SSD.

D-Verb Test

Whenever I’m reviewing a laptop I will perform the D-Verb test on it, as Pro Tools is my DAW of choice. You can do something similar in any DAW, but you may need to tweak a few things in order to perform this test. Many of the tools out there for stress testing are merely for the purpose of trying out overclocked components or fault-finding, so for me, there’s more value in a test that’s a little more in line with what musicians will be doing in the real world.

To start with, head to your Playback Engine in Pro Tools and make sure that 'Ignore Errors' is unchecked. You’ll then need to generate a sine wave utilising the signal generator in Pro Tools. To do this, you create a mono audio track, select five minutes on the timeline using the select tool, and then use the Audiosuite menu to render the signal. Once this is done, change the output of the track to Bus 1, then create a new audio track with its input as Bus 1.

Next, you need to add five instances of D-Verb to the audio track, then we can start duplicating. I recommend starting with 25-50 tracks initially, and you’ll need to make sure they’re all armed for recording, but don’t arm record your original track with the generated sine wave on it. Before you commence, make sure to open System Usage in the Windows menu so you can see what’s going on with your CPU during the test.

Once your tracks are armed make sure you turn your studio monitors and studio headphones down, then you just need to hit the record button. This will send your initially generated signal to all of the duplicated tracks, running it through all of the reverbs and recording it to track. Let it run for the full five minutes and listen back at low volume for any signs of crackles or pops. It’s pretty dull listening I’ll admit, but part of the test is the ability for your computer to run under load for a decent amount of time.

You can monitor what’s happening via the system usage window, which will let you know how much of the overall CPU is being used, as well as what the individual cores are doing. Provided your machine is able to handle it at 25-50 tracks, keep upping the stakes until you get Pro Tools to throw an error. On a recent MacBook M4, I got to over 400 tracks before I encountered my first error, which tells you what a modern laptop is capable of.

This test can be run on any DAW provided you don’t have a limited track count, and is a good way to see how much your computer can handle before you decide to take on a behemoth project. Pretty much every DAW has a tone generator built in, so whether you’re running in Ableton, FL Studio, Logic, or Studio One, you can perform the test. Any stock reverb will suffice, as most reverbs tend to be quite CPU-intensive. Most DAWs also allow you to monitor your system usage, you’ll just need to do a quick Google search on how exactly to do it in yours.

Stress Testing

If you've recently upgraded your machine or are trying to overclock to get more performance, I highly recommend using Prime95, which is available for Windows, Mac, and Linux machines. Originally designed to help find prime numbers, it’s become popular with those who build their own computers in recent years, providing a great benchmark to see how stable your computer is under heavy load.

The process of using it is very simple: just download the software, install it using administrator privileges, and when prompted, select the ‘Just stress testing’ option. The software will then run through a variety of tests and you’ll hear your fans fire up at certain points, as it also tests the cooling systems and power supplies in your computer.

You can run the test for any amount of time you like. The makers of Prime95 themselves recommend anywhere between 6-24 hours for optimal results, but if you don't have that amount of time you can run the test for however long you're typically using your computer. You can then check the results in the results.txt file that’ll be generated in the folder you extracted the program from.

One limitation of this test is that it’s primarily designed to detect faults, rather than simulate actual use. So if you own a newer laptop that is working fine you may not find out much doing it this way. If you’re overclocking your machine to extract more performance for your music-making however, or you’ve recently added more RAM, it’s a great way of seeing how much pressure you can put on your components and how stable the system is overall.

Conclusion

So there you have it, two relatively simple and free ways to see what your music-making machine is capable of. Of course, you can perform these tests on a desktop PC or even an iPad if that’s how you make your music, whether you’ve made a change recently, or you just want to find out just how far your computer can take your music production.

Wouldn’t it be great if every weekday of our lives we sprang out of bed, had a quick shower, opened the door to our home studio and got ourselves settled in for another full day completely immersed in our passion.

Alas, the reality is that unless you’re retired, or somehow able to avoid the 9-5 of work or school/college, music-making is something we have to find a way to fit into to our schedules.

Throw romantic relationships, families, children, exercise and general life admin into the mix and that window gets considerably thinner. This is all before we’ve even touched on countless other aspects of modern life, and various entertainment choices vying for our attention. Not to mention, of course, the temptation to just close your eyes and have a rest…

But, you’re a creative person, right? The act of making a new track gives you purpose. It's satisfying and life-affirming. It's why you do it.

If you can't squeeze in a music-making session, it’s highly probably you’ll feel frustrated, anxious and incredibly eager to get back into your musical zone as quickly as humanly possible.

It’s something that this very writer has definitely felt during particularly hectic weeks. New ideas for tracks or melodies almost taunting you with how frequently they pop into your brain when you’re miles away from where you want to be.

So then, once you’ve got that glorious two-hour stretch to actually work on some of your tunes, you want to walk away from the session feeling productive and positive, right?

Sometimes, that existing frustration can be elevated by the sheer inertia of endless choice, ruining advancing any further by the time it takes to simply try out different options.

That feeling of just constantly hitting walls when shaping up an arrangement. Or, spending too long kicking around the same track, so that you eventually lose all sense of perspective. It makes it very hard to finish anything satisfactorily.

Maybe auditioning another synth, trying another beat, or just notching out a frequency will suddenly bring the track closer to completion.

Not quite? Well, let’s try… oh, that’s your alarm going, time to unplug and head back to reality.

Maybe next time it’ll all come together…

The truth is, the music creation process is limitless, and your options are near-infinite. Similarly the mixing process is a delicate, painstaking art for even the most finely-honed of professional ears. Those of us sneaking it in between our 'adult' responsibilities will often just have to wing it as best we can if we ever want our music to be heard.

And, as we all know, winging-it is fraught with self-doubt.

There’s simply too much choice, and too little time.

It’s best then, to try and drain the overflowing pool, and impose a limit on all those options. Cutting down on the amount of tools you use, and the amount of time you spend on a track can shake things up.

Be your own boss, and force the change to happen.

Start by whittling down the options. With only two directions at the crossroads, rather than six (or 6,000), you'll be more trusting of your instincts, and not find yourself lost down one of those potential avenues of possibility.

Self-imposed limitations have been deployed by many of music’s chief innovators over the years. Not because they didn’t have time necessarily, but because studio self-control, or re-direction, can frequently push us out of our go-to comfort zones. And, they can be useful for nailing down more abstract, experimental pieces…

“When there’s a deadline, there’s also a destination, a context, a reason for something. And that’s what makes me finish [a track]. Up until that point, it’s an experiment,” Brian Eno told Red Bull Music Academy in 2013.

Eno’s infamous ‘Oblique Strategy’ cards were devised by him and multimedia artist Peter Schmidt back in 1975. They were designed to spur musicians to think outside of the orthodoxies of their practice (or their typical music-making routes) via triggering, open-to-interpretation instructions.

Some classic examples of these randomly selected cards include

- Cut a vital connection

- Turn it upside down

- What wouldn’t you do?

- Don’t be afraid of things because they’re easy to do

- Do we need holes?

The idea was to disrupt the typical thought-processes that musicians snap to under pressure, and suggest they approach their instruments differently.

What I’m talking about in this article however is similar to the Oblique Strategies, though a little less abstract. We're here to be more focussed on saving time as opposed to nudging you onto a different creative mindset.

Though, you might find yourself in uncharted territory just by the very nature of not doing what you’d normally do… and that's always fun.

We’ve previously explored this aspect of self-imposed rule-setting with a deep-dive on one of Eno’s most celebrated acquaintances, David Bowie, who himself was always hyper-aware of controlling his impulses in the studio. “Maybe I’ll write out five or six chords, then discipline myself to write something only with those five or six chords involved. So that particular dogma will dictate how the song is going to come out, not me and my sense of emotional self,” Bowie stated.

Let's boil it down then. Shrinking your production toolkit, instrumental choices and resisting overloading your mixes with 60+ tracks just slims the odds of wasting that precious (often hard-negotiated) time in your studio.

So, in the spirit of Eno’s Oblique Strategies (albeit a little less artsy) here’s a few of our own self-imposed rules you can try when starting your next track. Using these as a guide, you can probably concoct your own additional restrictions. Shake them up and choose two or three. Then apply them to your next track.

Hopefully, your process will be far speedier at least. And, you may find yourself surprised at the clarity of both the final track, and your creative mind.

1: No more than 10 tracks in the project

2: Keep the bassline repetitive throughout every part of the song

3: No more than three melodies

4: Make every track element with just one software synth

5: Absolutely no guitars

6: Absolutely nothing but guitars

7: No topline hook - just go for a mood

8: Ignore the temptation to swing into a chorus or drop, keep on a relentless verse

9: Do everything just using the DAW’s provided instruments and tools

10: Can you keep a one-chord tone/strum throughout? Try it

11: Only use loops

12: Once you finish your arrangement, remove two elements from it. Be brutal.

13: Use only one mixing tool per mixing aspect (one EQ, one compressor, one limiter etc)

14: Go completely drum/beat-free

15: Complete the song in one day (or one session - if you don’t have time)

Between devices like the Saturator, Pedal, Amp and recently-added Roar, Ableton Live has multiple tools that can be used to add distortion. While these might look fairly similar at first glance, these differing distortion tools can be used to create some very different effects, and can do more than simply rough-up sounds.

In our latest video tutorial, Ableton Certified Trainer Tom Glendinning explains a range of ways we can use distortion effects to creatively shape and mix sounds in Live. To hear these in action, he starts with a simple project making use of some drum variations, a sidechained bass part, chord stabs and a slightly wonky lead sound.

Tightening up drums with Saturator

The first element treated with distortion is a Drum Rack. Here Live’s Saturator effect is applied across the entire drum kit, which helps tighten up the drum sound and add a touch of extra weight. Saturator is used in its Analog Clip mode, which means it acts like a classic hardware limiter. As the Drive control is raised, it causes the peaks to distort slightly, which helps to both compress the drums slightly and add colouration.

Adding body to a kick with Pedal

Sticking with the drum parts, distortion is next applied to the kick drum. Here Live’s Pedal effect is used. This is a stompbox emulation that would commonly be used on guitars or synth lines, but it can sound fantastic on kicks too. By enabling the Sub option, we can distort our low-end kick without losing any of the bass frequencies. The Overdrive setting is used here with a medium amount of gain, which adds a nice analogue-style distortion that adds character and presence to the kick.

Distorted dub delays with Roar

Roar is the newest distortion effect in Live. Although it’s primarily used for adding grit and saturation, its routing and modulation options mean that it can be used for a lot more than just straight distortion. Here it’s applied to chord stabs to create a dub techno-style delay. This works by making use of Roar’s built-in feedback loop. By increasing the feedback time and amount, we can create echo-style delays that we can sync to project tempo. Best of all, we can apply distortion to just the feedback effect, which is great for gritty, characterful dub effects. The Shards mode works particularly nicely, creating sharp, crackly delays.

Pumping bass distortion with Road

Another appealing aspect of Roar’s design is the way that it can create rhythmic effects. Here we apply it to a bass part, and use the noise source in Roar’s modulation matrix to apply randomised modulation to the level of distortion being applied. Roar is then placed into Serial mode, to make use of two distortion lines. In the second stage, Roar applies additional distortion to our bass part triggered by a sidechain input from the drums.

Lead character with Erosion

The final distortion tool used in this example project is Live’s underrated Erosion device. This can apply distortion based on Noise, Wide Noise and Sine modes, which can be very characterful. Here we use Live’s LFO device in order to modulate the frequency of Erosion’s Wide Noise distortion, which adds some pleasing movement to the project’s lead sound.

DRUMS WEEK 2025: Many computer musicians shy away from programming their own acoustic (as opposed to electronic) drum tracks, understandably concerned that since they don’t know how to play a real drum kit, any attempt to create a performance on a virtual one will fall short in terms of technicality and sound.

Others go in the opposite direction, naively and over-confidently penning beats that at best would require five or six limbs to actually play, and at worst come across as mad and unconvincing, even to the untrained ear.

The fact is that programming realistic, groovy drum parts is well within any desktop producer’s abilities. All you need is a MIDI sequencer, a quality sound source and some insight into the rules, limitations and standard practices that real drummers work to and within.

But is it really worth the effort? Why not just use sampled loops of real drummers playing real drums to create your percussion parts? While this is always a wholly viable option, the results will be qualitatively different to what you’d get from building your own parts.

You would have to choose a loop based on its particular production and rhythmic qualities, and you’d have a very finite level of control over them. Program your own drums, though, and you have total command over every element of the ‘performance’ - from the notes themselves to the sounds of the individual drums and cymbals.

So, if you really want to take full ownership of your beats, building them yourself from the ground up is the way to go.

Same difference

If you’ve programmed electronic beats before, you’ll be glad to hear that the rhythmic paradigm that governs dance music and electronica is equally dominant in pop, rock and most other genres in which you’d find a live drummer. It’s that fabulous 4/4 backbeat, with the kick drum emphasising the first and third beats (either directly or by implication), and the snare on the second and fourth, embellished by incidental grace notes and fills.

The vast majority of modern music adheres to this straightforward structure, which is good news for programmers looking to emulate the real thing.

What’s more, knowing how drummers play their drums isn’t just useful for programming live-sounding parts: even your drum machine patterns will benefit from a true understanding of what the human ear has semi-consciously come to expect from contemporary percussion elements.

With all that in mind, then, it’s time to take a drum lesson…

Drumming 101  

The drum kit comprises a kick drum, a snare drum and a pair of hi-hats, plus a variable number of tom toms and cymbals (usually two or three toms and four or five cymbals, although the only limits are space and reach). Most of the action takes place on the kick, snare and hats, while the toms and cymbals are used for accenting and fills.

Most tracks will see the drummer spend the vast majority of their time playing a groove, which, generally speaking, involves nailing the backbeat on the kick and snare, and tapping out a metronomic eighth-note ‘tick’ on the hi-hats (known as ‘riding’).

Assuming the drummer is right-handed, the right foot plays the kick drum, while the left foot controls the hi-hat. The top hi-hat cymbal is struck with the right hand (crossing over the left), while the left hand hits the snare.

Try it now on a desk or your thighs: tap out eight evenly-spaced hits with your right hand, simultaneously tapping your right foot on hits 1 and 5 and your left hand on hits 3 and 7. Don’t worry about the left foot - just repeat ad nauseam. Congratulations – you’re drumming!

Feeling the groove 

Those are the fundamentals, but of course there’s much more to it than that. Rather than consciously calculating the placement of kick, snare and hi-hat notes while playing, the trained drummer ‘feels’ where they should go. He or she places them in the context of the rest of the track, in which - never forget - the drums play an underpinning, supporting role.

This isn’t something you need to master in order to program realistic drums, but you do need to understand the forms that these intuitive patterns might take beyond the elementary ‘kick, snare, kick, snare’.

Generally speaking, the kick drum should work in tandem with the bassline, so try placing kick hits directly on top of your main bass notes. Most of the time the snare will fall on beats 2 and 4, but shifting one forwards or backwards by half a beat can utterly transform a groove - try it.

Other variants are the half-time groove (à la dubstep), with the kick falling on beat 1 and the snare on beat 3. Then there’s the disco-style hi-hat pattern: you play 16th-notes with both of your hands, with the right hand moving to the snare for the backbeat.

Although it should go without saying, we’ll say it anyway: no drummer can hit more than two drums with their sticks at any one time. Just like us, they only have two arms! So, if you’ve programmed a snare, a tom and a hi-hat hit on the same beat, realism has left the building.

Little extras 

A couple of drum kit variables worth bearing in mind are the ride cymbal and double-kick drums. The ride cymbal is an alternative to the hi-hats. In pop and rock, it’s traditionally used in choruses and middle eights, where the energy of the track needs to increase. In jazz, it replaces the hats as the primary riding instrument – but jazz drumming is a whole other kettle of fish that we don’t have space to get into here. Another time…

Double-kick drums are found almost exclusively in the realm of metal, where they’re the bedrock for blisteringly fast double-time grooves and fills. Feel free to use them where appropriate, but be aware that they can quickly overpower a track, and that a drummer in the real world can’t easily pedal the hi-hat and play that second kick drum at the same time.

Laying down a basic groove 

Step 1: This is about as simple as a drum kit part gets: kick on beats 1 and 3, snare on 2 and 4, and hi-hats riding on every eighth-note. If this was the chorus of our track, we might move the hi-hat part to the ride cymbal. We’ve used the Ableton Session Dry kit but most DAWs will come with some decent acoustic kits to experiment with.

Step 2: Now we are upping the complexity a bit, so we draw in another kick drum note leading into the second half of the bar. You will hear what a big difference this seemingly minor change makes to the groove – although you still wouldn’t call it a complicated pattern.

Step 3: Displacing the first snare hit makes the groove become something else altogether. At the moment our drum part is authentic in that it doesn’t make any unreasonable demands on our ‘drummer’, but it’s way too rigid and programmed-sounding to pass for the real thing.

Dynamics and timing

As you’ll have no doubt noticed, our drum part so far, while making sense in terms of playability and pattern, still sounds totally mechanical and unrealistic. This is because we drew all the notes in by hand, snapped perfectly to the grid and at a constant, fixed velocity level.

The best way to remedy this problem is to play the part in live using a MIDI keyboard or drum pads. It does require a modicum of skill and the ability to ‘feel’ the groove rather than just rigidly bash it out, but you don’t have to play the whole thing at once.

Set up a record cycle and play each element in separately as the track loops round – hats, then kick, then snare, for example. The result will be a MIDI part with naturally varying velocities and subtle timing fluctuations from beat to beat (and they do need to be subtle – if anything sounds noticeably out of time, move it closer to the grid).

If you can’t or don’t want to play your part in live, draw the notes in with snap turned off, aiming for the grid lines (and being sure to fall within a few milliseconds of them). When you’re done entering the part, manually tweak your velocity levels or use a MIDI plugin to introduce a touch of randomisation/humanisation.

Keep it believable

Varying velocities and timing is the best way to make your parts sound realistic, then, but don’t go too mad! Keep your velocities within a relatively narrow range – a drummer will strive to be as consistent as possible, so obvious leaps in volume from hit to hit will sound wrong.

For hi-hat and ride cymbal patterns, the drummer will play the notes that fall on the beat harder than those falling in between (or vice-versa, if they’re looking to accent the off-beat).

Of course, it’s not enough to trigger just the one sample per drum at varying velocities – you need a properly multisampled drum kit, so that low-velocity hits trigger samples of drums played gently, while high-velocity hits fire off
full-strength recordings.

Your DAW most likely has a decent multisampled kit built in, and there are several fine third-party solutions: Toontrack’s EZdrummer and Superior Drummer, XLN Audio’s Addictive Drums, FXpansion’s BFD series, NI’s Battery and Studio Drummer are just some of the great titles available.

Finally, an important timing consideration is push/pull: skilled drummers will deliberately play very slightly behind or ahead of the beat to create a driving or laid-back feel. To emulate this, simply apply a bit of channel delay to the whole part, or manually shift selected elements of it after you’ve made all your other edits.

Getting real with timing and dynamics  

Step 1: Our drum part is currently triggering one of Ableton Live’s bundled drum kits, but you could try any of the third party options we mention in the text above. We experimented with some of the kits in Battery 4 for ours.

Step 2: We’ll load up our groove from the previous walkthrough in a moment, but first, here is our not so great attempt to play the same part in on a keyboard – one pass and no edits. Note the natural variations in timing and velocity. Well, we were after a more natural sound, but this might not be the ideal route for you…

Step 3: Let’s go back to our rigid, manually programmed part. The first thing we need to do is inject some human feel into our groove by deactivating snap and moving the hits slightly ‘off the grid’, emulating the intrinsic subtle variations in timing that you get with a real drummer. These edits need to be very subtle in order to work properly.

Step 4: Dynamics next, and we repeat Step 3 but for velocity rather than timing. Again, our adjustments are small – we want our snare drum hits, for example, to range from loud to slightly less loud, rather than loud to soft. We also want to make our off-beat hi-hats consistently quieter than the on-beat ones, as per the drumming norm.

Step 5: Shifting the hits back or forwards on the timeline to emulate the push or pull of a real drummer changes the feel of the whole track. Or you could introduce your own delay – apply 12ms of Track Delay to the drums to pull them slightly behind the beat.

Step 6: Alternatively, we can give our tune a more driving, energetic vibe by pushing the drums ahead of the beat – dialling in -30ms of Track Delay increases the momentum nicely. Generally, you can push your drums further than you can pull them before things just start to sound out of time.

Ghost notes, accents and fills  

Our groove sounds pretty authentic and ‘live’ so far, but there’s still more to do if we’re to make our drums sound like they’re being played by a real drummer with a real drummer’s brain.

To bring syncopation and a feeling of forward motion to the groove, we need to employ ghost notes. These are very quiet (often barely audible) incidental snare hits, funkily filling the spaces between hi-hat strokes. They can take the form of single hits or gentle presses, where the stick is pressed into the drumhead to create a very short, fast roll (three or four hits).

Applying ghost notes is an automatic, almost unconscious process for the experienced drummer, who will play them pretty much constantly. For the programmer who takes the time to draw or play them in, the sonic results will always be well worth the effort.

The snare drum is also used for accenting, with strategically placed full-strength hits adding character to the groove and/or emphasising other track elements. The classic example of this is the off-beat hit following the backbeat, or the drummer in a funk band rimshotting the snare in time with the brass section’s horn stabs.

Open hat strokes (whereby the left foot releases the hi-hat pedal, moving the two cymbals apart) serve a similar purpose, being deployed to lead into a bar or phrase, or doubled up with a snare hit to add bite and a burst of splashy sustain.

Fundamental fills 

Every eight or 16 bars, the drummer might throw in a fill, bringing the toms and crash cymbals into play. A fill is a roll around the toms, a series of enthusiastic snare hits or any other free-flowing deviation from the groove, usually concluding with a cymbal crash (which is always accompanied by a kick or snare – you’ll hardly ever hit a crash cymbal on its own). It’s designed to lead into the beginning of the following bar, which will serve a purpose in terms of phrasing - introducing a chorus or key change, say.

Fills can be simple but effective or complex and impressive, but all the usual programming rules apply: turn the snap off; vary the velocities, bearing in mind that the first hit in a series of strokes on one drum will naturally be played harder than those that follow; and don’t ever trigger more than two things (plus kick drum and hi-hat pedal) at once. If your sound source offers them, alternating between separate left- and right-hand samples can only boost realism.

Finally, fills don’t have to be full-on. There’s nothing wrong with a simple ‘rat-a-tat-tat’ on the snare - in fact, for many tracks, anything more than that will feel like overkill. Exercise restraint in your fill programming endeavours.

Adding ghost notes, accents and fills

Step 1: Ghost notes add movement and syncopation to any groove. We draw in some very low-velocity snare presses and single hits between certain hi-hat hits. To make the presses sound convincing, the first hit needs to be a bit louder than the following ones.

Step 2: Many drummers will play ghost notes between most of their hi-hat hits almost semi-consciously, so we add a few more. These really enhance the groove, and our drum part is now full of movement and sounding pretty funky.

Step 3: The only thing a real drummer would take issue with in our groove now is the lack of hi-hat variation. To remedy this, we place open hi-hats at the end of the first bar of our two-bar phrase and on top of the last snare. These are choked (that is, muted) by the following closed hi-hat hit – most virtual drum kits include this feature.

Step 4: For accenting and elaborating on the backbeat, extra full-strength snare hits are called for. These can be placed on or off the beat, depending on whether you’re aiming for syncopation or ‘beat reinforcement’. We go for syncopation most of the way here, with a big on-beat accent at the end of the phrase triggering a rimshot sample.

Step 5: Fills are used to mark the ends of phrases and lead into verses, choruses, middle eights, etc. Our first example is incredibly simple – the clichéd 16th-notes round the toms, with a cymbal smash at the end. We show you this for demonstration purposes only – don’t ever use it unless you’re specifically going for a naive, amateurish feel!

Step 6: This is the sort of thing a skilled drummer would actually play as a fill! We include a variety of snare articulations, different samples for left- and right-hand strokes and a set of Bernard Purdie-style hi-hat kicks (at maximum velocity). We’ve even thrown a tearing double-kick drum rip in there! We’ve drawn everything in by hand, with snap off.

DRUMS WEEK 2025: Drum kit instrument plugins are a wonderful thing, providing a vast array of realistic kit sounds and behaviours that you can use both with MIDI drum kit controllers or simply by programming in your DAW. They usually incorporate a MIDI pattern library, providing a great source of instant beats and fills.

However, if you want something a bit more specific, there’s no reason why you can’t program realistic parts from scratch. So whether you’re an okay drummer (like me) who wants to create more tricky parts they’d struggle to play, or a producer looking to create more realistic drum tracks, then read on as we take a look at some tips to get things moving.

Think like a drummer

1: This may sound pretty obvious, but we only have 4 limbs. So, as a drummer this means the most drum elements we can play at any one time is 4. Not only that, but 2 of those limbs can only play kick and hi hat foot pedal sounds, adding further limits to the kit pieces you can use simultaneously. So, when programming parts take into account these physical limitations and behaviours.

2: To give this some context, here we have a basic drum beat and you’ll see that for most of the pattern the kit pieces play individually or 2 at a time. Throw in a fill at bar 4 and briefly we’re playing 3 kit pieces simultaneously. There’s clearly not loads of drums playing together at any point.

3: This concept follows through into other aspects and a good example is crashes. Most drummers hitting a crash on the downbeat will use their hi hat hand and then move to the hi hats for the next hit. Here you can see this in the piano roll editor as the hi hat doesn’t play the downbeat.

Understanding your kit instrument

4: One reason drum kit instruments are so much better than bare samples is that they’re designed to include all the aspects of a multimiked kit, including close and distant mics, bleed between kit pieces and some variation (humanisation) in hits. However, they also often allow you to change these settings in ways that wouldn’t match a real kit.

5. For realistic parts it's important to retain some of this, or at least if you do change things have a valid reason. Here in Addictive Drums 2 for example, you can adjust the bleed level of individual kit pieces into the room and overheads. This can really help finesse the balance in these components. Here we’ve substantially reduced the hi hat level in the overheads.

6: The humanisation aspects of a drum kit instrument are also worth considering. For the most part these work well, but can sometimes be a bit too random or include obviously repetitive round robin implementation for example. Here we have activated the ‘No Alternating Samples’ option for the snare and used velocity to periodically change the timbre of strategic hits.

Focus on the hats

7: Drums are quite a physical instrument to play, and hitting things means you really feel the pulse of the music. This inevitably feeds through into the way you hit things and in particular hi hats and rides, creating natural accenting and dynamics.

8: One fast track route to making your kit parts more realistic is to build this concept into your programming. In this example the beat is very straight and we’re using higher velocity on the hi hat part to accent the kick and snare hits, just as a drummer would.

9: For patterns where the kick and snare don’t fall consistently on beats, we can use the hi hats to continue to mark the pulse of the music or divisions of that pulse. Here rather than accent each quarter note we’re accenting just beats 1 and 3.

Timing

10: Finally, real drumming is not quantized or even swing quantized, and although we may record to a click and even push the performance towards a quantize grid, we still want movement in the playing. To create realistic parts it’s important to factor this in, and even if you opt to quantize some of the quarter notes, building some feel elsewhere is vital.

11: The kick drum is responsible for much of the feel in a drum part, particularly if it’s playing grace notes. Playing with the timing of these 16ths and the immediate quarter note that follows can create a beat that either pushes or pulls. Here we’ve pushed both the 16th grace note and downbeat slightly earlier.

12: For snares even if the backbeat is tightly quantized we can add feel using ghost notes. Here our beat has 8th note hi hats, but adding in a 16th note snare ghost note on the second 16th of each bar creates more feel. We’ve used a very low velocity to make this hit super quiet. Try adding more ghost notes to taste.

DRUMS WEEK 2025: Here is a detailed, point-by-point checklist for approaching a standard drum kit recording session. First published a few years ago, it has been republished for MusicRadar's Drum Week

1. Skins, sticks and beaters 

It’s much easier to get volume, clarity and brightness from a new set of drumheads. If possible, get them on, tuned and ‘played in’ a day or two before the session.

Different types of head give different types of sound: single-ply heads are bright with less sustain (jazz/pop); two-ply heads are deeper and more controlled (rock/metal); coated heads are warmer in tone (disco/'70s). Your choice of sticks or hotrods and felt or wood kick drum beaters will also have a big effect on the sound of the kit. Get close to the sound you want before you think about microphones!

2. Tune up 

This should be the drummer’s responsibility, yet it’s more often than not the engineer or producer who realises that the drums sound wrong and knows what to do about it.

Here are a few tuning tips and things to check if you’re having problems.
Check that the head diameter is exactly right (fractionally larger than the drum) and that it’s mounted perfectly centrally. Always make tension adjustments in opposite pairs to avoid forcing the head off-centre. Seat the head by applying gentle pressure with your palm to the centre, then tune out any new wrinkles that appear.

3. Bottom heads on toms 

Wherever there are two heads, there are three possibilities. If you tune the bottom (resonator) head of a tom to the same pitch as the top (batter) head, you’ll get a purer tone with more sustain – perfect for jazz and its derivatives. If you tune the resonator lower, you get a slight pitch drop-off and less decay – great for rock.

Tuning the resonator higher shortens the decay further, which can be great for close miking. A good starting interval for the difference in pitch between heads is a minor third - ie, three semitones either way.

4. Front head on the kick or not? 

The front (resonator) skin on the kick drum is important for creating a good, meaty tone. Jazzers love it, and most rock/poppers have a hole cut in it to allow microphones and dampening material (a small blanket, say) to be placed inside.

For metal styles, where you’re mainly looking for click and thud, you may as well just remove the front head so that you can get a mic right in on the beater point.

5. Tighten the kick drum 

It’s possible to create a little kick drum chamber using, blankets, duvets or even a coat. These are draped over the kick drum, and any microphones are placed in and in front of it. Use a couple of small mic stands to hold the blankets up.

The chamber prevents spill from the cymbals and snare getting into the kick drum mics, making it easier to boost the high frequencies at the mix stage and thus enhance the ‘clickiness’ without adding harshness from the cymbals.

6. Playing to a click 

If you’re tracking to an existing recording, your drummer will need to be able to play to a click track. The best drummers are able to sit exactly on the click, but tend to flow around it in order to create a better feel.

Recording to a click makes it much easier to edit between drum takes at a later stage. If the whole band is tracking together, it’s best if only the drummer gets the click. Using a little beatbox or percussion groove usually makes a better ‘click’ than an actual click, and will also spill over into the microphones less.

7. Prepare the click for gaps 

Prepare for stops and gaps in the song by automating the level of the click in the drummer’s headphones. During the stops, lower the volume so that there’s no click spill into the drum mics from the headphones. If you’re recording the whole band, it’s also a good idea to send a click to the other players at that point, so the drummer doesn’t have to count everyone back in. This will save lots of painful editing and remedial work later on.

8. Get a crappy mic 

Tape recorders used to come with built-in condenser or electret microphones. Put one of these old machines into record/pause mode and take the output into your DAW for an exciting, super-crunchy sound.

Sometimes, rubbishy old battery-powered mics can sound good too. Put these types of mic in a place where they can hear the whole kit. The corners of the room can offer interesting results – experiment!

9. Record without cymbals 

Recording a kit in a difficult space without ideal equipment can be a fruitful and rewarding experience, but you might still feel that your results sound amateurish.

One of the classic drum sound problems is too much cymbal spill. This happens in the best of studios, and when it does, the pros just record the crashes afterwards. That way you can bring up all the exciting roominess of the drums themselves in the mix, then blend in perfectly recorded cymbals afterwards. As long as they share the same bus compressor no one will ever know (although your drummer will hate playing without cymbals).

10. Essential gubbins for a trouble-free drum session 

Moon Gel, for damping; 3-in-1 oil, for squeaky pedals; a drum key, because the drummer will forget theirs; an adjustable spanner, for rattly stands; gaffer tape, for holding stuff; masking tape, for attaching damping materials and the like; empty cigarette packets and J-Cloths, for that '70s sound; blankets; stage weight or sandbag; and Japanese cooking chopsticks - great for tom parts!

DRUMS WEEK 2025: Drums are often unfairly considered to be the poor cousin of the band. Cue the joke, ‘what do you call a person who hangs around with musicians? A drummer!’

But drummers have had the last laugh for quite a number of years. With the advent of drum machines and electronic kits, many sensible beatmakers, took to programming the machines and loops themselves. Why? Because they were the people that understood how patterns and grooves really worked, and that's long before we get into the topic of drum fills.

For 99.9% of all commercial music, the drums represent the core engine of a track. It's the element that drives the song along, and also the element that all musicians will listen to, in order to lock in to a groove.

Think about this logically; it’s the drummer that’s dictating the feel, and everyone else should listen and take note!

It's also important to stress that this sense of compliance is just as important at the DAW level, as it is for live work, but for producers and composers, having a basic understanding of how a drum kit actually works is invaluable to creating better drum parts and grooves.

The core components

Drum kits come in various shapes and sizes, and if we widen this to the world of production, we have many different colours and timbre-ranges for each instrument, along with a considerable number of elements which could be considered percussion instruments.

For our purposes, we’re going to keep things basic, distilling the drum kit down to 3 of its main components; the bass drum, the snare drum, and the hi-hat.

The bass drum (aka the kick) is the large drum which sits on the floor, and is played using a beater, connected to a pedal (usually played with the drummers' right-foot).

The snare drum consists of two drum skins above and below the drum (aka heads), with a set of springs (snares) which are stretched across the bottom drum head, creating that bright ‘rattly’ sound.

These two drums are thought to represent one of the most basic humanistic elements, used to create the rhythm of life! In this context, the kick represents the stamping of feet, while the snare reflects the clapping of hands.

If we bring this up-to-date and place it within the construct of a traditional musical bar of music, we end up with a kick drum on beats 1 & 3, and a snare drum on beats 2 & 4.

Conveying this to a drummer, notes are presented in notation form, through a specific form of drum notation.

Each line and space on a drum score stave, corresponds to a particular drum or drum kit element. Furthermore, drum kit notation uses a mixture of conventional and alternative note-heads.

The kick is written on the bottom space of a drum clef, while the snare drum is notated on the third space up. Importantly, the stems for these notes should go in a downward direction, while also using a conventional note head.

This basic construct is the absolute cornerstone of the vast majority of drum patterns. There are almost too many derivatives to cover in one article, but these can vary from adding further beats and accents, to displacing beats.

Even so, the drums will always be the element that sets up the basic feel and groove of your track, stemming from the kick and snare, in the first instance.

One easy and identifiable derivative of this basic pattern is to add the kick on all four beats of a bar. We could also add additional kick beats, or displace where the kick plays, as you can see in our example.

The third important element of the drum kit, subdivides the kick and snare patterns.

A physical hi-hat is made up of two cymbal-like components, that close together to form a unit. In this closed state, they produce a high and bright sound which is both percussive and relatively short.

Thanks to the use of a pedal system, which is normally operated by the drummers left foot, the two cymbals can be opened slightly, to create a sizzling sound, which is longer and more sustained. These two variations of sound are predictably described as closed and open hi-hat.

In its closed state, the hi-hat can provide a sense of drive and impetus, as it generally tends to play faster rhythms, sitting on top of the kick and snare patterns.

Interestingly, the type of pattern and speed, is often dictated by the style of music.

Similarly, certain styles of music rely heavily on the open hi-hat; it became very popular in the seventies to use an open hi-hat on the off-beats of a drum pattern, for disco and dance-based tracks. This influenced the dance culture of the 90s, and is still a drum staple today.

Switching the hi-hat part from an 1/8th-note pattern to a 1/16th-note pattern, will create quite a different feeling.

At certain points in musical history, it has become fashionable to place accents on certain beats, such as an accent on every third 16th-note, in the late 80s, or the classic funk grooves of the 70s and 90s. (In fact, many of the 70s grooves infiltrated the 90s by way of being sampled!)

Dictating your requirements for a hi-hat on a drum score is not complex, but it is important to be exacting.

Firstly, the hi-hat adopts a position which is just above the top line of the drum stave. In much the same way that the kick and snare note head stems go in a downward direction, the hi-hat stems must go in an upward direction.

This keeps the stems out of the way of each other, making it easier to read. However, it is also vital that when writing a hi-hat part, you use a ‘X’ as a note head.

This instructs the drummer to play the hi-hat, whereas using a conventional note head would indicate playing a tom instead.

By default, it will be taken as read that you want a closed hi-hat, but you can indicate your preference through the addition of ‘+’ and ‘o’ icons, above the note heads themselves.

The ‘+’ sign tells the drummer to play a closed hi-hat, whereas the ‘o’ indicates open, unsurprisingly!

One of the reasons that drummers became so involved in programming drum machines is because they understand the instrument, and it is a pretty safe bet that as they only deal in rhythmic matters, they will be far better placed to advise and help create a worthy groove.

This also extends to drum fills, which are the exciting bit you tend to hear at the end of every 8 or 16 bars, before you head into the next section of your song.

For this reason, it's become commonplace to provide a drummer with a basic score and concept of your requirements, before writing ‘slashes’ on your drum score, which indicates that the drummer should just get on and repeat your idea.

But when it comes the drum fills, just write the word ‘Fill’ above the drum score, where you would like your fill to be placed. In terms of fills, lose the shackles of the score and let your drummer have a little fun before returning to the fixed beat.

DRUMS WEEK 2025: Electronic drum sets offer the best in space-saving convenience, quiet practice and a diverse set of sounds right at our fingertips. But as a famous East Coast rapper probably once concurred, with mo’ features comes the potential for mo’ problems.

If you’ve got a head-scratcher of an issue that’s spoiling your enjoyment of your e-kit, check out our breakdown below where we detail some of the most common causes – and hopefully solutions – to your electronic drum kit problems.

Issue 1: “When I hit a pad, I hear other sounds coming from the module”

This is one of the most common issues with electronic drums, but thankfully it’s very easy to solve. ‘Crosstalk’ occurs when the sensors inside your pads pick up vibrations coming from striking another pad. The fix can be one of a few things. The first thing to check is that all of the pads are physically isolated. The chances are you’ll be able to hear which unwanted sound is triggering by mistake, but take a look around the kit and make sure none of your pads are touching each other. Likewise, vibrations can travel through your rack and mounts and cause other pads to trigger. Double-check that everything is mounted securely and all of the mounting hardware is sound.

If the problem is still occurring, it’s time to get under the hood with some module settings. Many modules have built-in crosstalk elimination controls, which will help the module to determine between an intentional strike and unwanted vibrations. Experiment with different degrees of crosstalk filtering and you should find that the problem goes away. However, depending on your sensitivity settings (see below) for each pad, you might find that one or more of your pads is set to register a strike at very low velocities.

Issue 2: “I have to hit the pads really hard to get a full sound, everything sounds like ghost-notes!”

When playing acoustic drums, we govern how loud a drum or cymbal sounds simply by hitting it harder or softer. The same is true for electronic drums, but our strikes are converted into an electronic scale rather than air pressure. Because of this, an electronic kit is likely to respond differently to any two players. Sometimes that could be subtle, or more extreme, but to combat this effect many modules are equipped with controls to help tailor the response to your playing. The sensitivity control is a great place to start, and can be adjusted to make the pad more or less sensitive to your stick striking the playing surface. You’ll need to dial it in to maintain dynamic response (so that you still get a variation between hard and soft sounds).

As well as the sensitivity setting, many modules are equipped with multiple velocity curve controls for each pad. This setting determines the power required from the player to transition between the softest soft and hardest hits, and can have a big impact on how your kit responds. Common velocity curves include: linear, logarithmic and exponential.

A Linear curve gives an even scale from the lightest to hardest strike, and is a typical ‘universal’ starting point. Logarithmic velocity curves tend to lend themselves to softer hits, requiring less effort to create a higher velocity strike. Conversely, an exponential velocity curve sees the jump from soft to harder strikes jump steeply. The result of this is that soft and medium-level hits are easily achievable, while the maximum-volume sounds will require more effort. An exponential curve is great if you’re a hard hitter and find yourself constantly triggering your pads at maximum velocity.

Issue 3: “Everything keeps moving!”

There’s nothing more frustrating than hitting a pad and watching it rotate with every strike. To fix this, give all of the fastened parts of your electronic drum kit a good check. Be careful not to overtighten things - you don’t want to break any of the mounting hardware, but it should be firmly secured. Some rack clamps include a finger-fastened bolt, as well as fixings that will require tightening with a drum key. If you’re setting up your kit for the first time, it’s worth spending a few minutes making sure that the frame is securely fastened. This includes checking that all of the supporting legs and crossbars are flat, even and level, and that the clamps are fixed in place with enough grip that they won’t rotate or slip when you’re ready to start hitting the pads.

The same applies if your kit has recently been moved, or has simply been left set up for a while without being checked. Get the frame solid at this point, and it’ll make your playing experience much more enjoyable.

Issue 4: “I’m not getting any sound from one of my pads!”

The first thing to check are your connections - your drum pad will most likely only have one socket for connecting to your module, so first, make sure that the cable is plugged into the pad properly. Still not working? Try swapping the jack from a pad that is to test it (be aware, the pad that you’re testing will now create the sound from the pad you’ve borrowed the cable from). If you hear sound, then you probably have a broken cable.

Some kits use individual jack cables to connect the pads to the kit. In this case, you can buy a new trigger cable and your pad will be back up and running. If your kit uses a cable loom or ‘snake’, the good news is that it’s replaceable, but the bad news is that you’ll have to replace the whole cable loom (these are available as replacement parts).

Alternatively, if you’re handy with a soldering iron, it’s possible to locate the break-point, cut the cable and attach a new one. We’d only suggest the latter if you (or someone you know) are confident in basic soldering, otherwise you risk damaging the other cables in the loom.

If testing the pad with a known-working cable doesn’t help, then unfortunately the pad itself might have developed a fault which requires repair, or worse, complete replacement. In this instance, it’s best to contact your kit’s manufacturer.

Issue 5: “My pedals are slipping around on the floor”

This is a common problem with electronic and acoustic drums, but when it comes to e-kits, the solution will depend on the type of hi-hat controller and bass drum pad you’re using. With an acoustic-style hi-hat stand, you should have rubber feet which are designed to grip the floor surface. While a hi-hat controller pedal will most likely be equipped with some short spikes to help it stay put. The same goes for your bass drum tower or shell. If you’re playing the kit on a hard floor, we’d advise placing your kit (or at least the pedals) on a rug, this way you won’t ruin your flooring. You can extend the spikes by turning the thumbscrew if you require a little more grip, just be careful not to go through the rug into your hard flooring, and remember that the spikes are there if you need to move the pedals. They’re unlikely to do major damage, but are often sharp enough to hurt and possibly draw blood if you grip them with your hand!

Issue 6: “My housemates complain about the noise, even though I’m wearing headphones!”

Noise is perhaps the drummer’s worst enemy, and while this might come as a surprise, hitting things with sticks and stomping on pedals is inevitably going to result in making noise.

Electronic drums go a long way to improve this, particularly if your kit is equipped with mesh-headed pads, but even then, you run the risk of annoying those around you by hitting pads hundreds of times just to play along with a song.

All is not lost, though. The first thing to work out is exactly what the problem is. If it’s simply that you’re playing at a time of day that causes irritation to your friends, family or even neighbours, a simple, cost-free solution could be to work out a better schedule.

If that’s not possible, though, there are a number of solutions. With the actual drum sounds confined to your headphones, it’s unlikely that it’s the sound you’re listening to is the problem. Rather, the airborne sound of your sticks hitting the pad, or the vibrations being transferred from your bass drum and hi-hat pedals and the kit’s rack are the culprit.

Airborne pad sounds tend not to travel too far, but can be reduced by using rubber stick tips, or possibly even dowel-style sticks to reduce the noise.

Vibration transfer is a slightly larger problem, particularly if you’re playing in an upstairs room. A popular DIY solution to killing the thud from your pads is to place your kit on a ‘tennis ball platform’. This requires building two platforms from wood, with tennis balls sandwiched between them to stop the vibrations reaching the floor.

If you’d prefer a pre-made solution, products such as Roland’s Noise Eater range are designed to absorb the vibrations from your stand and pedals, while its KDB-200 and KDB Quiet beaters can be fitted to any bass drum pedal, providing a reduction in impact noise.

Issue 7: “My pad has two zones, but I want different types of sounds on each, how do I do it?”

With a dual-zone pad, you’ll have a head and rim zone. These are completely separate, but share the same output, using a TRS (stereo) connection. Just because the pad is one physical unit, though, doesn’t mean you have to treat it as one.

Now, in the context of an acoustic-style drum kit preset, these zones are usually assigned within the module to give appropriate corresponding sounds. So, if it’s your snare drum pad, you’ll get a centre hit of the snare drum on the head zone, with a rimshot assigned to the rim zone. But, you might want (for example), a snare drum on the head zone and a hand clap on the rim. This is easily done, and it’s a case of telling the module which sound to play on each zone. Hit the zone you want to assign, then scroll through your module’s instrument list until you find the correct one. On Roland kits, you might need to unlink the Head and Rim - this feature is designed to pair different articulations of the same instrument to all zones on a pad, but in this case we want them to be treated like two individual single-zone triggers.

Issue 8: “How do I connect my electronic drum kit to a computer?”

First, it’s important to understand that there are two types of ‘recording’ when it comes to hooking your electronic drums up to a computer: Audio and MIDI. Audio is the recording of the sound that you play and hear in your headphones, while MIDI is captured as data which can be used to play back sounds from other equipment or software.

Depending on your kit’s module, you’ll either have a built-in interface (in the form of a USB connection) that can transmit and receive audio signals as well as MIDI, MIDI-only (over USB), or MIDI sent and received via traditional five-pin DIN sockets.

The first example means that you can take the correct USB cable, plug it into your module and connect the other end to your computer. From here, your kit will be recognised by DAW software such as GarageBand, Logic, Cubase, Ableton Live etc. as an audio and MIDI device. You’ll be able to send the sound of your kit into the software for recording, while also hearing any other tracks recorded in the software in your headphones while you play.

A MIDI interface built into your module will allow the transmission and receiving of MIDI data, so, you can play your kit, and the software will recognise the notes you’re playing along with controller data (for example, from your hi-hat pedal). MIDI is not sound, it’s a stream of instructions for MIDI-equipped musical instruments. So, instead of recording your playing as an audio sample, the software will use the MIDI information to play a plugin instrument, or back to your kit’s module. From here, the instrument receiving the MIDI data will interpret it, and trigger the corresponding sounds, which you will then hear in your headphones. This is how drummers are able to play impressive sampled sound libraries such as Toontrack Superior Drummer and BFD from affordable electronic drum sets.

If your kit is a bit older, it may not have a USB connection, and instead, MIDI will be provided via MIDI in/output sockets. In this case, you can still connect your kit to a computer, but you’ll need a little help from a third-party Audio/MIDI interface (we’d suggest looking for one that does both). This will allow you to record your sound straight out of your module’s audio outputs, while also giving you the facility for capturing MIDI if you want to play and record your performance as data for use with software sound libraries.

Issue 9: “How do I add more pads to my kit?”

Expanding your electronic kit is easier than you might think - in some cases, such as many Alesis electronic drum sets, the kit is shipped with a cable snake, which provides connections for all the included pads. But, a quick look at the back of the module will reveal additional sockets for adding extra pads. In these instances, it’s as simple as looking at your module, and hunting down the corresponding pad to connect. Alesis makes an all-in-one solution for its Nitro Max, which includes an additional tom and cymbal pad as well as the cables and mounting hardware you’ll need to add it into the kit.

A key point when it comes to adding pads to your kit (regardless of the brand) is the number of additional trigger inputs your module has, and how many zones they offer. It’s also a good idea to check the manual for your module (these are usually available on the manufacturer’s website) to confirm which pads are compatible with your module. In the case of digital pad-equipped Roland modules, it’s important to realise that any digital trigger you have connected to the module overrides the corresponding analogue trigger input. So, if you have Roland’s fantastic PD-140DS digital snare pad, you won’t be able to add another pad connected to the analogue connection labelled ‘snare’.

Issue 10: “How can I jam to music from Spotify with my electronic drum set?”

We should point out that this tip isn’t just for Spotify - it will work with any sound that your playback device is capable of generating. If your module has Bluetooth, you can hook it up wirelessly to a phone, tablet, computer or any other Bluetooth audio player. The method for this will be different depending on your module, but usually you can begin pairing the module to your device via a Bluetooth button, or if not, switching Bluetooth on in the module’s setting.

Next, you’ll need to go into your playback device’s Bluetooth menu and allow the connection. Once it’s setup, you should hear the music in your headphones which are connected to the drum module.

If you don’t have Bluetooth, there’s every chance that your module has an auxiliary input. These normally use a 3.5mm mini-jack connection, so you’ll need to buy a cable.

Look for a stereo connection. Assuming you’re plugging it into a phone, tablet or headphone output from your laptop then you’ll want a stereo mini-jack cable. Remember that little adapter that came with your phone? You’ll need that, as most phones don’t have a dedicated headphone output, and instead can send audio through the charging port. For hifi’s or CD players, check your output options - it might be that you need an RCA (phono) to mini-jack cable.

DRUMS WEEK 2025: Record drums? Why would you bother when it’s so easy to simply launch Superior Drummer or BFD, bash out a quick beat on the keyboard, hit the quantise button and get stuck into recording everything else?

It’s certainly true that some of these virtual drums/drummers sound awesome, and offer a mind-boggling choice of sounds and patterns.

Not only that, but they don’t argue, and they don’t issue demands for payment, writing credits or bowls of Smarties with all the blue ones taken out – so it’s effectively a no-brainer, right?

Well, here are some other things that those pieces of drum software – and all the rest of them – don’t do. They don’t know how your song goes; they don’t have ideas; they don’t have a background of musical influences; they don’t know how to make your songs come to life; and they don’t sound unique.

If you’ve ever spent hours trying to program a fill that sounds right, or hi-hat parts that groove, or grace notes on a snare, or a convincing crash-ride, then you’ll know that for all their amazingness, these superb virtual instruments require a lot of work to produce totally lifelike results.

But isn’t recording drums a total nightmare? No, that’s a myth – recording drums is actually surprisingly easy. Admittedly, it’s often the sound of a badly recorded kit that makes an otherwise great track sound like a ‘demo’, but if you follow some simple rules, you can avoid the pitfalls.

Of course, a big, posh recording studio would make things easier, but we’re going to assume you haven’t got one of those and show you how to turn your DAW into a drum studio with nothing more than a basic set of microphones, an eight-channel audio interface, an extra set of headphones and a splitter box.

Apart from the recording gear, the drum kit and the room, there is, of course, one more vital component: the drummer! If your drummer’s not up to scratch, all your efforts will have been wasted. There’s no point listening to a great recording of something crap – but there’s definitely a point in listening to a crap recording of something that’s great! So make sure your drummer is experienced, knows how to tune their kit, hits well and consistently, and can play with a click.

Preparing to record

When setting up a recording session, the first thing you need to sort out is the room in which the drums are going to be recorded. The question is: live or dead?

If you’re making a rock track, you’ll need a ‘live’-sounding room or an adjacent live-sounding room where you can put ambient mics. If there’s a rug on a wooden floor, get rid of it – drums always sound better on a hard surface.

For a dry, disco-sounding kit, you’ll need a ‘dead’ room, like the lounge or bedroom, where soft furnishings help to calm the sound.

The drum kit

A good drummer will know how important it is to get the kit sounding right. Freshly tuned, new heads will have a crisp attack and make the drums nice and loud. Loud drums make the room sound better and don’t get drowned out by cymbal wash.

A common mistake people make is over-damping the drums in pursuit of a dry, '70s type of sound. Bear in mind that lots of top-end resonance and room ambience gets absorbed in the mix by guitars – especially in rock music. Over-damping also makes the drums quiet, so easy does it!

Keep a selection of blankets, pillows and bricks around to help get the kick drum right. Try and keep some life in the sound; don’t muffle it so much that it sounds like a cardboard box. Use the dampening to just shorten the sound. The best thing for dampening the top head of the snare is Moon Gel by RTOM. It costs about £8 and makes fine-tuning snare sounds easy.

The session

Before you hit the record button, you need to prepare your session. Working with a sprawling production featuring 20 tracks of backing vocals and tons of plugins is going to mean complications down the line.

By printing or bouncing down the key elements of your track as new audio files, you can unload all plugins from the session. This will prevent latency problems, so that the drummer will hear everything right in his headphones and there won’t be any complicated latency compensation processes happening in the background.

You should aim to end up with a few bounced ‘stems’ – guide drums, bass, guitars, keys, vocals. Make sure they all start on bar 3 or later, so that you have at least a couple of bars of count-in. The click or guide drum loop can be MIDI-triggered or printed as audio.

These few tracks then need to be turned down to around -15dB – this will leave plenty of headroom for the click to be clearly heard by the drummer over the backing track and the drums.

Getting the spot mics right

Step 1: Once you’ve got your kick sounding good, put a dynamic cardioid mic, like the AKG D112 shown here, inside it. The closer it is to the beater contact point, the punchier a sound you’ll get – 5-6" away is about right, at a 45º angle to the batter (top) head. A small boom stand is ideal for this.

Step 2: This next step isn’t essential, but is well worth trying. Take another mic (dynamic or condenser) and place it outside the kick – this results in a more rounded and meaty sound. Don’t get any closer than 5-6" to the skin, though. Weight both kick drum mic stands with bricks so they don’t flop around.

Step 3: Use a dynamic cardioid mic, like this Shure SM57, for the snare batter head. Try to have it looking in and down onto the centre of the head. You’ll need to keep it outside or near the edge of the drum so that it doesn’t get in the drummer’s way, but this will sound better anyway. As an alternative, try having the mic looking horizontally across the snare.

Step 4: Again not essential, but a mic under the snare can help to bring out the rattle of the snares, as well as improving the fatness of the overall sound. If your overheads are good (see next page), you may end up using more of this in the mix than the top snare channel. If your audio interface has a phase reverse button, activate it for this mic’s channel.

Overheads

The most important microphones in our whole setup are the two overheads, which basically record the whole kit sound. Ideally, you’ll have a pair of large diaphragm condenser mics on tall stands two or three feet above the kit and about four feet apart. These are a ‘spaced pair’ of omni-directionals or cardioids. Omnis will give you more of the room sound, while cardioids achieve more focus.

Although this setup is common practice and can sound great, it’s not actually the easiest to get right, and it can also sound terrible.

The main problem is phase – you need to ensure that the sounds from your drums get to all of the overheads at roughly the same time. The more mics you use, the more complicated the phase relationships get and the less focused the sound can become, leading to a loss of punch and low frequencies in the recording.

Google the XY, ORTF and MS overhead miking techniques, all of which give very different sounding results – and all with just two condenser mics. Experiment to find out which suits your kit and room.

The Glyn Johns method

Mega-producer/engineer Glyn Johns has recorded many of the greats. Here’s his quick and economical kit miking method, which has become something of a standard over the years:

• Overhead 1 (cardioid condenser, panned left) is about 4-6" above the floor tom, facing the hi-hats across the snare.
• Overhead 2 (cardioid condenser, panned right) is about 2-3' above the kit, pointing down between the toms and snare
• Snare spot mic
• Kick spot mic.

Room mics, bottom snare mic and second kick mic are all optional. This method does depend on having a great-sounding kit and room, though, so it could be a bit unflattering to a home setup.

Below we’re going to use the two-mic overhead technique that’s become known as the ‘Recorderman’. It involves quite a close mic placement, and minimises the room sound in the overheads. Here’s how it’s done…

Getting the ambient mics right

Step 1: For overhead 1, mount a large diaphragm cardioid condenser (if possible) looking down on the snare drum from a height of roughly two drumsticks. This is a starting point that can be adjusted to tweak cymbal and room spill. This technique will give a tight sound, focused on the snare – to tighten the sound further, you could set a hyper-cardioid pattern.

Step 2: The second condenser (OH2) looks directly at the snare over the drummer’s right shoulder. To set the distance, make a loop of string starting at the centre of the kick, going up to the top overhead mic, looping round your finger, and then down to the centre of the snare. Keep the snare end in position and bring your finger back towards the second mic.

Step 3: In a dead or bad-sounding room, the ambience won’t be great, but it’s useful to set up an ambience mic some distance from the kit anyway, as we’ll see later. Even better is putting a mic up in an adjacent space that’s bright-sounding, like a bathroom, kitchen or corridor. This signal can be delayed to create the illusion of size.

Step 4: Even though we’ll be getting most of our sound from the overheads and kick mics, it’s a good idea to put dynamic mics on the toms in the same way we did the top of the snare. If nothing else, these can be used to trigger tom samples if your tom sound isn’t that great.

Getting a great recording

Step 1: Plug the mics into your audio interface. The overheads go to inputs 1 and 2 and will need 48v phantom power (unless you’re using ribbon mics – in which case, don’t touch that switch!). Kick in and out go to inputs 3 and 4, while snare top and bottom go to 5 and 6. The ambient mic (hopefully a condenser, so requiring phantom power) goes into 7, and the floor tom mic into 8.

Step 2: Get the drummer to beat the hell out of the kit. They always play louder on the actual takes, so don’t worry about getting too hot a signal – having the highest peaks spike at around -5dB is a good target. There’s something to be gained from clipping, but not on the way into the interface – keep it clean.

Step 3: Once your levels are sorted, it’s a good idea to run a test take to check for silly mistakes and any uninvited hum, hiss, clicks or crackles. You can also use the test recording to check the phase of the mics and ensure a fully fat sound on playback. Even though the theory of phase is complex, the practice is easy…

Step 4: Insert a trim or gain plugin into each of your eight drum channels. Bring up the first kick mic, followed by the second. When the two are balanced, try inverting the phase on the second kick channel (the left/right Phase Invert icons). Leave the button set where the sound is best.

Step 5: Now turn down the kicks and bring up the overheads, panned centre. Try inverting the phase of one of them. If the overheads sound fatter with just one channel phase inverted, leave it like that. Bring the kicks back in and blend them with the overheads. Change the phase on both the overheads now, and leave them set at wherever the kick sounds fattest.

Step 6: Balance in the top snare mic and switch the phase to whichever setting sounds fattest with the kick and overheads. Bring up the bottom snare mic and set the phase to the opposite of whatever the top is set to. Repeat this process with the floor tom and ambient mics. This is quick ’n’ dirty, but now you’re ready to record.

Quiet studio, please

You should now be ready to press the record button, safe in the knowledge that you have the sound of the drums completely nailed. Of course, there’s a whole bunch of stuff to do post recording to turn the raw drum takes into a mixed kit, but we’ll cover that in future features. First, though, we have to make sure we record everything we need.

There’s nothing worse than putting up your drum recordings, long after the drummer has packed up and left, only to find that the bottom snare mic hasn’t recorded or that your middle section has click track from the headphones spilling all over it. That’s why a test take is important; here are a few tips on how to get the most out of the session.

Setting up a rough live mix

Step 1: Although you won’t create your final drum sound until the mixdown, you’ll need to get your mix half-decent for recording. Here’s some CPU-friendly advice to help you get it right. Create two auxiliary sends and returns from the ambience track. Insert a delay plugin over the first return and set the L and R Delay times to 30ms and 60ms.

Step 2: Next, insert a room reverb over the second aux return set to around 0.6s Decay time. Compress this and add a harmonic distortion (we’ve used the Phatr FX again). This artificial space will help the kit sound more ‘finished’ while you’re tracking – and a more exciting sound will make your drummer play better.

Step 3: Take your individual drum channels and the two new aux returns and set their outputs to a new bus. Create a new aux return for this bus and call it ‘Kit master’. This is your main drum channel. Now create two or three more busses and returns for the rest of your recorded track stems (guitars, bass, vocals, etc).

Step 4: You should now have five or six ‘master’ faders, making it easy to create a good working balance and hear the drums properly in context. You can also create a separate sub-mix for the drummer by adding aux sends to the master faders and routing them discretely to the headphones output. To minimise latency, keep your audio interface buffer size as small as possible.

DRUMS WEEK 2025: You wouldn't be the first person to think that recreating a realistic sounding drum part should be an easy task in today's day and age, thanks to the immense technology that we have at our fingertips. The reality however can provide great challenges, with stodgy drum tracks that don't sound anything like real players. And, some listeners can really, really tell.

Even if the equipment in your arsenal is based around the samples that come with your DAW, there is still plenty you can do to gain the perfect groove. In the first instance, it's incredibly important to remember that you are trying to create something that sounds realistic, and there are several key elements which will stand you in good stead.

Firstly, we have to talk about feel.

The degree to which ‘feel' may influence your drum programming, tends to centre around the genre of music you are aiming to recreate. Commercial styles of music, particularly those associated with pop or dance, are likely to invoke heavier degrees of quantisation. Styles of music such as jazz and funk, are far more dependent upon an individual drummer’s performance and their style of playing. Consequently, it's going to be more difficult, but there are still things that you can do.

Always listen to examples of the music which you are trying to recreate in the drum sphere. Ask yourself what each drum element is doing and how it informs the groove. This can often be broken down into three instrumental elements, within the drum kit; the kick drum, snare and hi-hat.

One tell-tale giveaway often relates to the style of playing in the hi-hat. You’ll want to avoid the ‘machine-gun-like’ repetition that you find in so many programmed hi-hat patterns, in favour of adding accents on certain beats of the bar (or even offbeats) along with the occasional open hi-hat.

Within the realm of MIDI, the hardness with which you hit a note or drum pad registers on a scale between 0-127, as a recorded value known as the ‘velocity’ of a note.

Armed with some decent drum samples, you will find that these are often multi-layered, so hitting a note harder will not only sound louder, but will also provide a slightly different timbre to the sound. It should replicate the organic instrument.

It's a very similar procedure for both kick drum and snare drum parts, however the latter will often benefit from occasional ghost notes; these are much quieter notes that add colour to a drum groove, outside of the much louder backbeat.

Imagine that your snare part is loud on beats 2 & 4, but you are playing much quieter notes at other points within the bar.

These quieter notes are known as 'ghost' notes, and are particularly prevalent in more complex music, such as funk and jazz, and can very commonly be found in rock styles of music too.

How well these ghost notes audibly perform, may be indicated by the complexity of your available samples, and indeed the type of snare drum that has been sampled.

When trying to create a part which sounds humanistic, it's important to consider the player's perspective. If a drummer is hitting a kick drum, snare drum and hi-hat all at the same time, it stands to reason they might not land at literally the exact same time.

If you find yourself using quantise, you will find these instruments trigger at exactly the same time, which will undoubtedly yield a machine-like performance. Apply a small degree of humanise, which will stagger where each note lands, but in a very subtle way. The softening of the heavy transient, created by all three instruments triggering simultaneously, will fool the ear in to thinking that there’s more humanistic behaviour in the track than there actually is.

While we are talking about human performances, remember that a human being only has so many limbs, and in most drum examples, can only hit three drum elements at the same time. (There are exceptions to this, such as double-bass drums, but it’s a safe bet!) If your drum part extends beyond this number, it's going to be an impossible drum part, which is likely to give the game away!

We all succumb to the mercy of the plugin, and we are all guilty of adjusting values on screen to suit the numerics in our head.

There's only one way to tell if your MIDI drum performance sounds realistic, and that's to take your eyes off the screen, and listen objectively to the part you have created. Be truthful, and ask yourself if it really does sound like a real drummer. If it doesn't, try to ascertain the elements which are still not quite right, but always be truthful about the outcome, because if you can hear it, the chances are other listeners will too!

Programming drums using samples and MIDI triggering is not for everyone, but thankfully there are a substantial number of very worthy plugins that can help you in this pursuit. Better still, many of these started life with drummers playing live grooves, and it's unlikely that there is a better way to get a realistic outcome.

Here are some of our top recommendations

Toontrack EZdrummer 3

Toontrack is arguably the company that started the ball rolling for those wanting to integrate live-sounding MIDI-based drums into their songs. That was over 20 years ago, but the company has not stood still for one moment, producing new and ever enhanced versions of its popular software.

The entry-level EZDrummer 3 is a deceptive example; populated with a staggering number of samples, across 7 drum kits, collated by legendary producer Michael Ibert, and at the equally legendary Hansa Studios in Berlin. The multi-layered sampling is enhanced by bespoke kit mixes, recorded in different rooms, within the Hansa studio complex. The package is enhanced by a considerable collection of very usable MIDI templates, triggered and played by a professional drummer, with percussion examples too.

There is also a substantial Toontrack expansion sample and MIDI library, allowing you to build upon the package, zoning in on the style of music you want to make. You can even build drum tracks within the package and sync it to your DAW.

If all this sounds too good to be true, you wait until you see what EZBass and EZKeys can respectively do, in a very similar way, linking together these essential components for a realistic sounding band within your DAW

UJam - Virtual Drummer

UJam’s take on MIDI drums requires the user to trigger loops and styles manually from the keyboard. The advantage here is that you can select and change different grooves on-the-fly, but it can take a degree of practice to get something to sound coherent.

There are many different volumes in the UJam library, which can be extended to build up a sizeable collection of different drum sounds, across many different styles.

XLN Audio - Addictive Drums 2

Addictive Drums 2 is another healthy suite of extensive acoustic drum samples, accompanied by MIDI grooves and expansion possibilities. The MIDI element is more basic than some other packages, but XLN has gone the extra mile with the interface, which is particularly appealing to use.

There are various package options from the outset, although like all of these plugins, completing the collection could be a costly pursuit!

Toontrack - Superior Drummer 3

Building upon the superb success of EZDrummer, Superior Drummer 3 took things up a considerable notch, with over 230GB of samples, an enormous library of live MIDI drums (curated by style and song section, with alternatives and fills) and a comprehensive mixer section, which allows access to a colossal number of microphone channels, used for the recording of each kit.

The basic package (of which there is nothing basic) is engineered by legendary production mastermind George Massenburg, and is completely backward compatible with EZDrummer content. However, the package really comes into its own through the SDX expansion libraries, which offer some of the absolute best drum samples available today, recorded in some of the best drum rooms in the world.

Whether you want a kit for metal, or lighter jazz duties, Toontrack has you covered. Even the most seasoned musician will find it difficult to ascertain whether or not it's the real thing, because in essence, it is!

Comparatively, it's not cheap, but in our view, it is the best.

Whether you're working from your DAW samples or a third-party suite, one final element to consider is the glue that binds the drum track together, at the backend of the mix.

Speak to any studio engineer, and they will have their favourite option for drum compression. Placing a compressor across the drum bus will have a very positive effect on the overall output, from any plugin or indeed live drummer.

Unsurprisingly, there are many vintage-in-software designs that would do a fine job, and in-tune with other suggestions we have made, always use your ears to form your own opinion. If you still need some inspiration though, try a basic 1176-style FET compressor, a Neve 33609, or one of our favourites, the API 2500 plugin, available from Universal Audio.

Remember, use your ears… and be objective!

DRUMS WEEK 2025: For as long as human beings have walked the earth, there have been drums – and for almost as long, they’ve been accompanied by cymbals! From hollowed-out logs and frame-mounted hides through all manner of vaudevillian, military and orchestral percussion paraphernalia, the history of drumming is epic and diverse.

This tale writ large and loud ultimately culminates in the modern drum kit, a seemingly disparate collection of instruments that came together at the start of the 20th century and forever changed the face of music.

From jazz and rock ’n’ roll, through pop, rock, soul, country and funk, and on to hip-hop, house, techno, drum ’n’ bass and every other genre of the last 80 or so years, the drum kit, whether real, synthesised or sampled, has been a constant presence. And although many of the playing techniques and styles have changed a lot over the decades, the essential make-up of the drum kit has remained largely the same, centring on the kick drum, snare drum and hi-hats/ride cymbal, supplemented by tom toms and crash cymbals.

If you’ve spent any time hanging around drummers, you may have heard them refer to their drum kit as comprising a number of ‘pieces’.

This only refers to the number of drums in the kit – cymbals aren’t counted. And with the kick and snare being absolutely essential, the rest of the pieces are usually tom toms: a four-piece kit will have one mounted on or suspended over the kick drum – also known as a rack tom – and one floor tom, while a five-piece will add an extra rack tom (or possibly a second kick drum).

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Beyond that, there are really no limits, and any number of rack toms, floor toms, rototoms, octobans, concert toms, gong drums or anything else the drummer fancies can be added into the mix.

As touched on a moment ago, many drummers also employ a second kick drum, although space-saving double bass drum pedals are available as an alternative, enabling the same functionality to be had from the one drum.

Metal works

As for the cymbals, beyond the fundamental requirement of a pair of hi-hats and a ride, there really is no upper or lower limit to the amount of metalwork a drummer might choose to hang around the kit – space and reach permitting, of course.

Some drummers like to envelop themselves in veritable clouds of the things, while others are satisfied with a couple of crashes. If you're looking for examples of some really over-the-top setups, try Googling the drum kits of Terry Bozzio, Mike Portnoy and Neil Peart. They might look kind of ridiculous, but their big-name owners absolutely make use of every component.

Today’s sample-based virtual drum kits pack all of these elements into a flexible software interface for triggering by keyboard, drum pads, manual note entry and, in most cases, an included library of MIDI clips.

Throughout the following pages in this month's set of features dedicated to the beat, we’re going to be dealing with both real and virtual drum kits. Our step-by-step walkthroughs will show you how to record the real thing, program and humanise the sample-based alternatives, and process, mix and enhance both. Let’s get started!

And what of the drum machines?

This issue is primarily dedicated to recorded and sampled acoustic drums, but (of course) no programmer of beats should overlook the characterful, sonically expansive sounds of the synthetic drum machine.

With a history going all the way back to the 30s, the drum machine has gone through endless changes, fashions, declines and revivals, ultimately finding its natural home in dance and electronic music.

Although the sounds made by these electronic boxes might be miles away from their acoustic counterparts, all of them still categorise their sounds as regular drum kit components – kick, snare, toms, hi-hats, etc – and they’re often programmed to play exactly the same sorts of parts that real drummers play, or at the very least interpretations of them.

It’s testament to the longevity and continued relevance of the classics – most notably the Roland TR range – that many of the software plugin drum machines we use in our DAWs today are emulations of them.

It’s not all nostalgia, though: there are also plenty of completely original virtual drum synths on the market, taking the concept to amazing new heights. Some of the best drum apps we've looked at recently include Sugar Bytes DrumComputer, Accusonus Rhythmiq, Mixed In Key Captain Beat and Auddict PercX Pro.

Breaking down the drum kit

Hi-hats

A pair of 12-15" cymbals mounted on a stand and brought together by pressing down a pedal at the base, the hi-hats can be played with sticks (with the cymbals closed most of the time, but often half-open for a ‘splashy’ sound) or pedalled to provide a distinctive ‘chick’ sound.

The most commonly used playing technique is with the right hand (assuming the drummer is right-handed) ‘riding’ the hats on eighth-notes, crossing over the left hand, which strikes the snare.

In some styles (most notably disco), the hi-hats play 16th-notes using both hands, with the right hand moving to the snare for the backbeat.

Tom toms

AKA rack toms. Used to provide syncopated embellishment to grooves and pseudo-melodic meat to fills and solos, tom toms range from 8" to 16" in diameter and are tuned to a specific set of pitches that also incorporates the floor tom(s) at the lower end.

Crash cymbals

Crashes are thin cymbals, generally ranging in size from 8" to 18", with anything below 13" referred to as a splash.

They’re used for accenting significant beats and marking changes in a track, such as the transition into the chorus, and are almost invariably struck at the same time as the kick or snare.

Variations on the crash cymbal include the China type, the crash-ride and, er, the Rocktagon – Google it.

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Ride cymbal

A large (20-24"), heavy cymbal struck with the tip of the drumstick on the body, or the shoulder of the stick on the bell (the bulging section in the centre).

While the majority of eighth-note riding action takes place on the hi-hats, for choruses and middle eights, drummers often switch to the ride cymbal for its bright, floaty, sustained sound.

The bell can also be struck for a shorter, louder, more cutting sound that’s typically associated with rock but is thoroughly usable in any genre.

Floor tom

Essentially extra-deep tom toms on legs, floor toms range in diameter from 14-18" and sit next to and behind the kick drum. They’re quite simply an extension of the rack toms, providing a deep bottom end to the tom tom frequency range.

Snare drum

A shallow (between 3" and 6.5" deep, usually) drum with a rack of ‘crimped’ springs stretched across the bottom head, the snare provides that cornerstone of popular music known as the backbeat: relentless hits on beats 2 and 4.

The snare can be struck on the head alone, or on the head and rim at the same time – this is a complicated-sounding manoeuvre called a rimshot that’s actually an almost unconscious technique for any half-decent drummer.

The sidestick (often erroneously also called a rimshot), meanwhile, involves holding the tip of the stick on the centre of the head and dropping the butt end down onto the rim. This produces the distinctive ‘click’ that’s been the mainstay of many an '80s ballad.

Kick drum

The kick drum (or bass drum, as it’s traditionally known) delivers the power and bottom end of the drum kit. Measuring between 18 and 24" in diameter, and sitting on its side on the floor, it’s played with the foot using a pedal-mounted beater.

In pop, rock and many styles of dance music, the kick drum generally complements the bassline, directly following it or weaving around it. In house, techno and other four-to-the-floor styles, of course, it simply nails every beat of the bar – surely the most ubiquitous and powerful rhythmic device of the last 30 years.

Many metal drummers use two bass drums for the fast double-kick grooves that are essential to that style.

Auto Filter is easily one of our most used devices in Ableton Live. It's long been a workhorse tool ideal for shaping sounds, blending elements in and out of arrangements and manipulating tracks for live performances. As of the recent Live 12.2 update, it's received a massive glow-up with multiple new filter models, LFO options and more.

In the first of a new series of tutorials, Ableton Certified Trainer Tom Glendinning walks us through several ways these new filter functions can be used to add character and movement to a variety of track elements. To demonstrate this, Tom uses a track sketch inspired by classic French house composed of drums, bass, keys and a funky guitar loop.

Comb Filter

The first track element that Auto-Filter is applied to is a keys loop. The new Comb Filter mode is used to create a phasing effect that runs through the chord pattern. This is only really audible when the filter moves through the frequency range, so Auto Filter’s LFO is used to modulate the cutoff. By increasing the Phase Offset option, we get variations in the cutoff position between the left and right channels, adding extra stereo width.

Resampling Filter

A second instance of Auto Filter is also added to the keys track. This time it’s set to Resampling mode. This creates an effect similar to bitcrushing, emulating the low bitrate tone of old school samplers. This is set with a cutoff of 12kHz to create a subtle retro ‘ringing’ tone.

Vowel Filter

Next Auto Filter is applied to the track’s bass part. This time the new Vowel filter mode is used to create a vocal-like tone that makes the bass sound funkier and more retro. Auto Filter’s envelope follower is used here to create cutoff modulation that follows the volume level of the incoming bass signal. The LFO is used for additional movement, and a little Drive is added for a touch of grittiness.

Side-Chained Filter

Another Auto Filter is added to the end of the bass effect chain. This is left in the default low pass mode, but the sidechain functionality is used so that the cutoff frequency responds to the level of a secondary input, in this case the keys. This creates a rhythmic effect where the bassline fades in and out as the chords play.

Notch+LP Filter

The next element is a guitar loop. Here Auto Filter is used with the Notch+LP mode engaged, another new addition with Live 12.2. This filter mode combines a low-pass filter rolling off the top end frequencies with a more surgical notch filter lower down the spectrum. Here a little resonance bump is applied along with some modulation via the envelope follower and LFO, which creates a more funky and vibrant effect. The LFO is placed in Spin mode, which creates an effect where the guitar moves around the stereo field.

DJ Filter

Finally another Auto Filter is added to the master channel. This is used in DJ mode, which applies low- and high-pass filtering depending on the direction in which the cutoff is adjusted. This is perfect for live performance or for creating arrangements, allowing us to filter out frequencies as a track builds towards a drop.

Tom Glendinning is a session musician, sound engineer, sound designer, music producer, composer, and VJ. As an Ableton-certified trainer he helps musicians be creative with Ableton Live. Connect with him at: https://elphnt.io/

PLUGIN WEEK 2025: Analogue emulations make up a large component of the plugin landscape with options ranging from compressors and EQs to delays, reverbs and tape machines.

If we want to create a truly representative emulation of a fully analogue recording and mixing system, then we need to chain these plugins together in some way.

In this tutorial we’ll show you how to create a classic analogue style mixing chain from playback through to the mix bus. We’ll look at the type of plugins to use and consider signal flow options and how to deal with signal levels.

1. Let’s start by assuming our production consists of audio we have recorded directly via our audio interface or sourced from software instruments. In essence none of it has been through a tape based recording system, so it should have low noise and no tape saturation.

2. In a classic analogue recording system, signals are recorded and then played back from tape, and this is therefore the first element of our chain. Even so, tape emulations impact many aspects of a sound and add varying amounts of distortion, and this may not be what you want. We’ll leave that decision with you, but this is one aspect of the chain you might want to bypass for some sounds.

3. When we use tape emulations in our DAW, the plugin creates a real time emulation of both the recording and playback processes. This emulation often includes choice of tape, tape speed, tape bias, pre/post EQ and some form of operating level calibration, which affects the level of magnetic flux applied to the tape. Try different tapes and calibrations to find what you like. In general a modern tape (499 or GP9) running at 30IPS will be quite clean while older tape types (456 or 250) at 15IPS will be more coloured

4. For tape we set input levels to match the type of sound. In the analogue domain, this is an art in itself, but a general rule would be to aim for 0dB peak on the tape VU meter unless the sound has obvious transients, like kicks and snares. Here, start at around -10dB VU on peaks. Whatever the source sound, listen critically to the tape compression and adjust the input level to get the sound you want.

5. The next stage is the main channel strip, which consists of a line amplifier stage followed by EQ and compression. You have a couple of approaches here. One is to select component plugins to taste, and the other is to use a channel strip plugin. Using individual plugins provides more flexible signal flow and allows you to insert more powerful digital processors. Here we’ve chained together a Neve style EQ and dbx compressor, with a more precise digital EQ plugin in between.

6. Using a dedicated channel strip plugin is less flexible, as if you want to add more EQs, compressors and so on you can only stick them either side of the channel strip plugin. However, dedicated channel strip plugins sometimes have quite handy features to help you replicate mixing desk behaviours.

7. On our bass sound we’re using the Brainworx Focusrite SC channel, which is a highly regarded analogue emulation that includes various ways to maximise the analogue aspects. The V Gain at the top adjusts the noise level, and the THD screw adjusts the saturation. Meanwhile, the TMT option introduces subtle differences between plugin instances, so if you use the Focusrite SC plugin across multiple channels you’ll have an outcome more like an analogue console.

8. Whichever route you take, most analogue emulation plugins include some form of level dependent saturation behaviour. The operating levels at play are not always obvious as not all plugin developers tell you. Even so, it’s important to be aware that if you have a chain of analogue emulation plugins and you turn the output up high on one then the next plugin may saturate more than you expect.

9. Mix bus saturation is a key aspect of analogue desks and although you could use a regular channel strip emulation in this role, there are some specific options, including Slate’s Virtual Console, and various options from Sonimus including Satson, which emulates a classic SSL mix bus. Place this simple plugin in your mix bus for additional harmonic distortion and left/right crosstalk.

10. One final decision you might want to make is to use tape as the mastering medium. This will add a further tape EQ curve, low level saturation and some cross talk. In this case you’ll want a stereo format tape emulation (¼” or ½”) as the final mix bus insert. Here we’ve loaded up Slate’s Virtual Tape Machine in its ½” mode with the lowest noise tape formulation, setting the input levels so the meters peak around 0dB VU. Try adjusting the input level to find the amount of desired tape compression.

PLUGIN WEEK 2025: Reverb is a hugely versatile tool that can be used to create all sorts of effects and transitions. Here, we're going to show you how to use reverb to create filters, reverses, risers and pads...

Let’s start by creating some interesting filter effects. Import a drum loop or other sound source into a new DAW project, insert an algorithmic reverb, and set its wet/dry balance to 50%. Choice of reverb has a big impact on both the method and results here, so you may need to experiment to get the effect to work.

What we are essentially trying to do is create a controlled feedback situation. A small room/reverb size combined with longer decay times tends to do it, and some processors also have a feedback setting which does the trick nicely!

Pre-delay can have an influence here, too. In short, experiment with parameters and plugins, and you’ll find a whole bunch of interesting filter effects.

Let’s make a real reverse reverb now. Create a new project and import a vocal or other lead part. Insert a reverb on the vocal and set up a pleasing, medium-to-long effect. When happy, use your DAW’s audio processing tools to reverse the reverb part – not all DAWs can do this, so use a free audio editor such as Audacity if needs be.

Import the bounced audio to a new track if your DAW didn’t do this automatically, then reverse it once again (or, if using Audacity for processing, reverse and then import to your DAW). Carefully position the reversed bounced part so that it lines up properly with the unreversed original part, then mute that dry vocal’s track. You now have true reverse reverb!

Risers use similar techniques to reverse reverb, but tend to be less of a faff. Open a project and find a point in the song that would benefit from being preceded by a riser. Export a short segment of audio from this point – sufficient to seed the reverb, and including any tracks that you want to include in the riser sound.

Create a new project and import the snippet you just exported. Insert a reverb, set it to 100% wet, and create a sound with a long tail and plenty of character. Add other plugins for more sonic interest if needed. When done, export the results, import this new file back to the original project, reverse it and position it.

Let’s now create a synth pad. Open a new project and create a new instrument track using a synth or sampler that supports granular playback. We’re going to use Arturia Pigments, but you could use NI Kontakt, Steinberg Padshop, BLEASS SampleWiz 2, or any one of numerous premium and free granular synths should you not already have something suitable.

Import into your instrument into the unreversed riser sound you created in step 5. Use the synth’s sample and granular editing controls to hone in on a pleasing portion of the source sound. Build up a patch for the sound by adding a pad-like amp envelope, and modulating the granular position time to create timbre shifts.

Finally, let’s play with some freezing. Import a drum loop into a new project and insert a reverb that has a freeze function – we’re going to use BLEASS Reverb. Set up a nice ’verb, add an automation track for the freeze button, and create a rhythmic automation pattern. You could then enhance the effect by automating other parameters.

PLUGIN WEEK 2025: Our goal with our recent batch of 'Start Here' features has been to help you make and release stellar music, armed with just your computer and DAW, even if you have no prior experience. Across these features, we aim to detail every aspect of music production in as clear a way as possible, with straightforward advice and tips to help you make better music.

So far, we've explored MIDI, Interfaces, recording external synths and much more. In process terms, we have introduced the DAW - the Digital Audio Workstation - at the heart of computer music making, we have detailed how to start a track, arrange a track with loops and also how to finish a track.

This time around we're looking at mixing a track and how you can make your ideas sound better with good mix practice for the instruments, vocals, beats and more so that everything in your song shines!

Luckily, DAWs come with fantastic mixing features, including faders to adjust volumes of instruments and mix effects like EQ and compressors to tweak the sounds.

As with a lot of things when it comes to music production, being a little restrained yields benefits, so not only will we explain what to have in your mix, but what to avoid - less is often more when it comes to mixing, and small tweaks here can mean big results there.

Whatever music you make, though, learning how to mix it is almost as important as hitting on your musical idea in the first place. A good mix will put that idea on show, whereas a bad mix could well bury it forever. Before we go into the details of mixing, we now explain what the process is from its first principles.

What is mixing?

What is mixing?

When you mix your music you are simply making sure that all of your instruments, ideas, melodies, vocals and beats can be heard without hindering one another.

On a very basic level, this means making sure that certain tracks aren't too quiet while others aren't too loud and drowning out everything else.

Mixing means letting instrument tracks breathe, giving them space where required and certainly not letting them clip - that is when the audio pushes the visual feedback into the red on your virtual mixing desk.

As well as volume considerations, mixing is about compression and EQ. The former is making sure that signal levels remain even; you might have a bass guitar track that is quiet in some places but loud in others.

A compressor allows you to boost the quiet parts and compress the loud ones so you get a more even signal. EQ on the other hand is all about taming different frequencies, maybe cutting the low end of that wild bass synth so it's not too muddy, or boosting the mid frequencies of a vocal so that it cuts through a mix and can be heard better.

Finally, there's the concept of panning. In a stereo mix (there are surround mixes, but we're focussing on the more standard left and right stereo channels here) you have a stereo soundstage in which to place your instruments, which not only lets you keep clashing instruments away from one another but can offer a lovely, wide experience for your listener. We'll tackle all three of these areas – volume, EQ/compression, and panning in more detail below.

Why do you need to know about mixing?

Why do you need to know about mixing?

We'll assume that you have a few ideas either looping in your DAW over several audio and MIDI tracks, or you have a fairly advanced arranged project ready to mix.

The first thing to do is to set all of your fader levels at 0dB which on most DAWs leaves you some headroom to go up, although we'll more likely mix downwards so 0dB will be your peak. Some people advocate -6dB to give yourself even more headroom but for the sake or our workshop, 0dB is just fine.

The main principles, then, are to be able to hear every track at first. You can do this by adjusting fader levels.

Pull them down from 0dB rather than up to keep that headroom - the important thing is that channels must not clip or go into the red part of each mixer channel.

You will probably find that even by adjusting levels, some sounds will clash, as everything so far is playing through the centre of your stereo soundstage, and that's where panning comes into play.

Panning simply lets you spread your instruments and vocals across a stereo spread from left to right.

While it's really up to you where you place instruments and vocals, there are some rules to follow, should you wish. One good piece of advice when mixing a live instrument/band-based piece of music (anything with traditional guitars, bass, drums and vocals) is to picture the band on stage and pan the relative positions of the instruments in that way, so drums and bass in the middle, guitars panned slightly left and right; maybe backing vocals panned to one side too.

Generally speaking, low frequency instruments - your kick drum and bas guitar, for example - stay central as does you lead vocal, but otherwise, fill your boots and experiment.

The one mixing rule that you should abide by is that 'if it sounds good, it is good', even if you break all of the other rules!

With compression, you can use it creatively to crush a sound and get some filthy distorted beats, for example, but in a mixing scenario, it is used more subtly to keep levels even. It's especially good for wayward vocals, guitar, bass and acoustic drums, where playing might not be even and some notes and beats might be louder than others.

A compressor literally compresses the higher volume sounds so that they are more even with the lower, at the same time boosting quieter notes so that they are more of a match for the louder ones. You do this by setting a threshold, a level that you want your volume to be, and the compressor will lower any signals above this at a speed defined by the attack value which controls how quickly the compression takes place and a release value, which is how quickly it stops.

On a basic level, EQ is used to boost the frequencies you want to hear and cut those you don't, but it's not just a matter of adjusting bass and treble as all of your instruments will sit in specific frequency ranges within the 20Hz to 20 kHz range of human hearing.

Broadly speaking, these ranges and instruments are as follows: sub bass is 20-60Hz, the lowest end of human hearing and often difficult to hear unless you own a subwoofer, but particularly important in bass heavy dance music.

The bass range is 60Hz to around 250Hz and covers your bass guitar, some male vocals, and parts of your kick drum.

The lower mid frequency range is next, covering 250Hz to 500Hz and takes in guitars, vocals, and some bass guitar elements. The mid frequency range is around 500Hz to 2kHz and deals with a lot of vocals, some percussion and keyboards.

The upper-mid range goes from 2kHz to 4kHz and takes in account female vocals, snares and strings, and finally the higher frequencies (4kHz and upwards) deal with very high percussive frequencies like cymbals.

We'll cover some of the main EQ rules in our key points on mixing below but generally the idea is to use EQ to emphasise what you want to hear by boosting certain frequency ranges, the width of which is defined by a Q value, and cut the frequencies you don't want to hear.

Overall it can sometimes be useful to think of a mix in three dimensions: panning left and right, volume up and down and EQ/compression front to back, and really the idea is to spread a mix so it fills a canvas. Use some of our tips below to help you get your mix picture perfect but remember rules are only a guid line, and if it sounds good…

10 key mixing points

10 key mixing points

1. Volume and panning are probably your first mix practices and just by getting these two right you should be well on your way to creating a good mix. Remember with the former not to mix too loud as it will be easy to end up pushing everything upwards to compete. It's often best to pull back channels to emphasise others rather than pushing everything up.

2. Generally speaking, mix decisions are about subtle tweaks here and there rather than dramatic edits. You don't want big volume differences between tracks and nor do you want extreme panning unless you are a more experimental composer. Similarly over-boosting and cutting certain frequencies might well muddy your mix so don't overdo it.

3. And taking this point further, always mix at low levels. Keep you maximum headroom at 0dB but also your overall volume level down. You might end up mixing for long sessions and you can easily damage your ears mixing at high levels. Sure, you can blast it out on occasion but keep the volume levels low for the bulk of the session.

4. When you mix, it's good to have an EQ and compressor on every channel. You could and maybe should have this set up in your DAW template so that whenever you start a track, all of your channels are set up and ready to go. This is particularly useful for some mix engineers who advocate cutting the low frequencies on every channels bar the bass synth, bass guitar and kick drum - using a high pass filter on an EQ - as these are really the only channels which should be using low frequencies and every other channel might have stray frequencies in this range which muddy the bass.

5. Inevitably, some instruments will clash when panning, a very common one being both the kick and bass if they are both placed centrally, and this is where EQ comes into its own, because it too can be used as a separator. For the kick and bass, you can use use the EQ's UI to see which parts of the frequency range are clashing and use a band pass or bell shape to boost this clashing area in one part and cut it in the other, perhaps nudging up frequencies around the cut to keep the instrument's presence in the mix.

6. One EQ rule worth following is 'boost wide and cut narrow' so keep the Q value larger when boosting certain ranges and low when cutting. This is because cutting is often used to remove problem frequencies, like keyboard or guitar body noise and ringing, so needs to be surgically narrow to avoid degrading more of the original recording. Wider boosts, on the other hand, sound more natural than narrow boosts which can sound overly dramatic.

7. If after your EQ-ing, your individual track sounds too degraded when you solo it, don't worry, it is all about how it sounds in a mix. You might well have cut the body out of a drum loop, for example, so that it sits well with a bass guitar. Yes it sounds weedy when you solo it, but as long as it sounds good in the context of a mix, you're good to go!

8. Modern DAWs make it very tempting to ladle on track after track when mixing, but even if you are a master of panning, EQ and compression, sooner or later you will end up with a massive wall of sound and a muddy mix. Again, restraint is often the option, so pull back the track count and let your initial ideas shine brighter and you will make more impact. Take a leaf out of Finneas and Billy Eilish's book; they sold 10 million copies of Eilish's debut album with songs with often just four instruments on. (Ok they might have had hundreds of vocal tracks, but you know what we mean!)

9. Use a reference track to mix. Load in an audio file of a finished track by your favourite artist, in the genre you are producing within, and try to 'mix up' to it. That is, try and reach the same dynamic range as your reference track to get the same kinds of peaks and troughs. Don't expect to reach the volume levels as that is done in the mastering stage (which we will cover in an upcoming Start Here) but a reference track will give you a good indication of the kinds of level differences and EQ targets you should be aiming for.

10. And finally the good old tip that we can pretty much put in any mixing or production feature: try your finished mix out on different playback systems - car, bluetooth system, headphones etc - to make sure it sounds good on all of them.

If, for example, it sounds consistently bass-light on these systems, it might suggest that the monitors or headphones you are using to mix are too bass-heavy and you are overcompensating by pulling back the bass when mixing. If so, you'll need to adjust the bass on your monitors downwards. Most have controls to do this.

Further reading

MusicRadar further reading

First off, here's 9 quick and effective ways to improve your mix

Here's some guidance on making your mix sound more pro-level

Want some guidance on setting up some solid mixing chain templates that will be perfect for a range of music styles? Look no further.

Sometimes it pays to defer to a smarter set of ears when handling some of mixing's more problematic challenges. While they don't have ears, these plugins and tools can certainly bring some on-the-fly know-how to improve your mixing - and save your precious time.

We asked some professional mixers to dispense some of their many years of experience in tip form, here's what they shared.

Getting vocals right and to sit comfortably in a mix can be a real challenge. Here's a route you can apply to your own mix templates to capture and mix a lead vocal effectively

While you're here, you may as well grab a selection of the very best free mixing plugins you should certainly add to your arsenal. Here's 6 of our recommended ones.

Mixing and mastering are processes that feed off and out of each other, to that end much of the wisdom pertaining to one directly affects the other. Here's a selection of mixing and mastering techniques every bedroom producer should know

Beats can prove a particularly troublesome spot to get right in the mix, here's an approach we recommend.

While a technical process, mixing very much depends on the ears, intuition and emotional responses of a human. Or does it? Here's where we put a new AI mixing-assistant to task and square it off against a real human mix engineer

PLUGIN WEEK 2025: The sample buffer is one of the dark arts of the digital audio domain. Ever got up to a microphone, started to record a vocal take, and found yourself hearing a delayed signal coming back at you? Or, perhaps you are playing a virtual instrument, with an impressive delay between the triggering of a note, and the sound emitting from your speakers.

Both of these scenarios relate to the sample buffer, but with some management, these issues can be improved or even eliminated!

The sample buffer is a setting which is controlled by the user, designed to influence the amount of time being set aside for your computer to process audio. It affects the signal as it enters your system, normally via some form of audio interface, before the processed audio is returned to you for your monitoring pleasure.

Your computer is doing an awful lot of processing while recording digitally. As your audio is inputted or processed, your computer is fighting furiously to crunch the signal into digits.

This number-crunching takes a short amount of time, creating a delay to the signal which is described as latency.

This can be managed, but regrettably, you are never going to completely eliminate latency from your system. You can also employ workarounds to achieve the best results.

One day of course, computing technology will be so powerful, latency will be negligible - but we are still some way off from this benchmark.

The amount of induced delay in the signal, is assigned via the ‘sample buffer size’ setting. You will find this setting in one of your DAW’s setting or preference panes, described as something like ‘I/O Buffer Size’, with an accompanying drop-down menu relating to the number of samples being set aside for the processing buffering.

Most systems will offer you half a dozen or so numerics, ranging from 32 samples up to 1024 samples. The lower this number, the lower the amount of latency, whereas a higher number will induce large amounts of latency.

In real terms, this means that the lower value is affording less time for the computer to process the data, where as a larger number affords the computer to more time.

So which option is best? It could be tempting to choose the lowest buffer size value available, which will in theory generate the least amount of latency. That is undoubtedly the case, but in reality you are likely to experience instability. This could range from glitching in your audio, to your DAW exclaiming that it does not have enough processing power, before bringing your song to a shuddering halt. In the worst case scenario, the DAW might even crash!

It is therefore in your interest to select a value which strikes a happy balance between latency, and giving your computer less reasons to grumble.

Knowing which buffer size to select depends on a couple of factors. The primary influences will be your package and the speed of your computer. More up-to-date/faster computers, will be able to cope with a lower buffer size, but before you rush to the computer store to purchase the most up-to-date machine you can afford, the size of your project will also affect how the buffering performs.

When you start a project, you are likely to spend most of your time tracking-on, either from audio or instruments in the software domain. It therefore makes sense to set your buffer size as small as your computer can handle, without interfering with your working practice.

As your project increases in size, and hopefully nears completion, you will likely enter the phase where you begin to think more about your mix. At this stage, you may add more plugs and effects to the project, meaning that there is more audio processing for your computer to undertake. So it makes sense to increase the buffer size at this stage, so that your computer can handle the increased processing of data. The short delay between hitting the ‘play’ button and hearing your track start, will be largely unnoticeable.

As we hinted earlier, there are also some workaround solutions which may help your cause. One of the most useful and effective relates to the recording of audio.

When you purchased your audio interface, you were probably directed towards the company website to download the latest drivers for your product.

As part of this process, you may have been advised to use a software product alongside your DAW, which looks a little like a mixing desk.

Thanks to some clever software engineering, most companies have a solution which provides an exceptionally low latency signal path, working alongside your DAW.

This means that you can monitor your live audio through the interface software, as you record, without the annoying delay coming back at you.

You will however have to mute the ‘live’ channel in your DAW, otherwise you’ll hear the signal twice!

Normally, the accompanying software just springs into life, the moment you start work, and it's a good idea to embrace this working practice, as you’ll achieve better results using this 2-step solution.

Meanwhile, if you find yourself working with virtual instrument plugins, you are entirely at the mercy of the buffer size.

This can be immensely annoying, particularly when working with larger orchestral sample libraries, which are renowned for inducing large amounts of latency, especially in complex legato string patches.

One trick-of-the-trade is to employ a keyboard which has an inbuilt sound engine.

By engaging an internal sound from your keyboard (such as a basic piano) it will act as a musical counterbalance to your delayed sample, providing musical and aural feedback which makes tracking orchestral or latency-heavy patches far easier.

It's not ideal, but it’s a useful and workable solution!