I’m totally happy to say that I produce predominantly ‘in the box’ i.e. using a DAW and digital tools. Sure, I record vocals, guitars and other instruments with an external interface and preamps occasionally, but all of my writing and mixing is done inside of Logic.
However, one of the biggest issues that we face as digital producers is getting that special nuance and warmth that analog processing and instruments can provide.
Which is better, analog sound or digital?
I’ve been fortunate to work in a number of studios and experience the tactile, tangible nature of analog sound equipment. Is it great? Yes. Is it better than digital tools? Personally, I find this to be a moot point.
At present, each have their own advantages and disadvantages, and we have the ability to utilise the strengths of both. So why not?
While many people state that you simply can’t beat hardware, it’s not impossible to achieve a very similar sound using only software…and you can get darn close. On the contrary, software has many advantages over analog gear, and I believe it’s certainly possible to get the best of both worlds.
One of the biggest problems with digital production is that it’s simply too good. That sounds like a strange thing to say, but it’s absolutely true. Often you’ll hear people complain that music sounds ‘too digital’ – this is what they’re referring to.One of the biggest problems with digital production is that it's simply too good.Click To Tweet
Digital recording and instruments can often come across as ‘cold’ and flat because they’re essentially ‘too perfect’. Analog hardware and synthesisers exhibit anomalies and errors that have become synonymous with their sound.
Moreover, often these ‘flaws’ are pleasing to the ear, generating warmth, excitement and unpredictability. There’s a reason that companies like Universal Audio go to incredibly great lengths to replicate these analog discrepancies within their software – analog imperfections sound better to the human ear.
Is this by design or through years of conditioning? No one knows for sure, but there are plenty of theories online if you care to look!
For now, though, I’d like to discuss a number of the techniques and processes that I use to achieve a more analog sound in the box.
There are countless ways of achieving this and, while I can’t cover them all (never mind the fact that I’m constantly discovering new techniques myself), I’ve tried to list some of my main go-to methods in this post.
By the end of this post, you’ll discover how to do the following:
- List 5 techniques for creating an analog sound;
- Understand how they can manipulate the source;
- Apply these techniques using a variety of tools;
- Evaluate how effective each of these methods is in augmenting your own music.
I hope to expand on these techniques in the future, and where possible I’ll update this article to include my latest discoveries.
[Note: In this article I occasionally recommend plugins from third party providers – please note that I am not affiliated with these companies and I don’t receive any commission. I’m simply suggesting their tools based on my experience using them and their reputation.]
Noise is one of your best friends in music production. It can be used to create dramatic transitions, to calibrate meters and it can even reveal flaws in your acoustic environment.
One of the most creative uses for white noise is in combination with other sounds. It’s a great alternative to distortion, as it can give the impression of a subtle saturation, without losing the power of the original sound.
In addition, generating noise separately to the main signal provides a greater amount of flexibility as you can blend the sounds with more control. For example, you may have a saw wave that could use some extra grit. In this case, you can generate white noise on a separate synth and mix it back in with the original to taste.
Noise is also very useful for enhancing drum sounds, and even building them from scratch. For example, you may have a snare sample that has the perfect amount of body, but it lacks top-end or ‘fuzz’ to cut through the mix.
You can layer in some bright white noise to improve the overall brightness, without affecting the main sound with EQ.
Many synthesiser software instruments have a noise oscillator built in. Xfer’s Serum, for example, has a separate noise sampler with a variety of options for different timbres of noise:
Real instruments and analog synths sound ‘real’ for a reason. One of the biggest variables is pitch. Subtle pitch deviations occur when instruments perform notes, and our brains automatically associate these anomalies with the sound of real instruments.
Pitch variations, glides and bends can all be recreated to add interest to digital instruments.
Many analog synthesisers even require tuning, and the pitch can drift depending on conditions such as weather, humidity and temperature, much like any other acoustic instrument.
You can mimic these anomalous pitch changes digitally using several methods:
a) Pitch bend
You can control pitch bend range directly within most software instruments and synths. As an example, I’ll use the well known third party synthesiser Serum from Xfer again.
An ideal bend range is 12 semitones, as this spans an entire octave and will be easy to hear.
Serum allows you to individually set the bend range both upwards and downwards, so here I’ve gone with 12 semitones each way for a spread of 2 octaves in total. This provides a lot of flexibility for performance.
Once you have MIDI notes recorded on your software instrument, open up the MIDI region and then open the MIDI Draw pane (or equivalent). Depending on what DAW you’re using, this may be located in a different place.
Open up the pitch bend pane and you’re now able to draw in any custom pitch bend modulation that you’d like to implement on your melody, chords, drone and so on.
b) Note glide
Closely related to pitch bend is note glide. Note glide refers to the degree of drift between two pitches when played consecutively.
So, for example, if I play C and jump up to G, you’ll hear a certain amount of pitch drift that can be pre-determined within the instrument’s controls.
In Logic’s Retrosynth, for example, you can directly control note glide from the main screen. It’s worthwhile recording in your notes and playing around with the amount of drift – even a subtle amount can create a much more natural and interesting sound.
Pitch drifts have become particularly popular in the last couple of years. Disclosure adopt this technique to create an old-school, more analog vibe reminiscent of tape fluctuation.
An example of this can be found in Omen featuring Sam Smith. The pitch modulation is most obvious during the bridge starting at 37 seconds.
c) LFO pitch modulation
On top of the above methods, you can also manipulate pitch using an LFO (‘Low Frequency Oscillator’).
Many synths have a built-in LFO but, even if they don’t, there are many free-standing plugins that enable you to apply LFO manipulation separately.
In the following image, I’ve created an LFO envelope in Serum and set it up to modulate the pitch of Oscillator 1 as shown:
You can also use separate tools such as Xfer’s LFO Tool and Logic’s AutoFilter to modulate pitch. Be sure to check your DAW’s stock tools as it’s likely you already have an LFO filter at hand.
MIDI modules, too, such as Logic’s MIDI FX Modifier and Modulator (you used to have to set these kind of effects up using Logic’s Environment), enable you to manipulate multiple instrument parameters via MIDI CC signals, and can therefore be used to alter pitch.
When used subtly, this can have the effect of generating a more analog sound, imitating the pitch variations of analog synths.
Simply manipulating dynamics is one of the most effective ways to introduce a more natural and ‘organic’ feeling to your instrumentation.
When playing an instrument, controlling dynamics is one of the key ways that musicians are able to interpret a piece, express themselves and convey an emotion.
Therefore, if you can incorporate expressive use of dynamic changes within your music, it is much more likely to engage and connect with your listener.
It’s also impossible for a human being to perform notes at exactly the same dynamic level consistently.
Lack of dynamic variation is perhaps the single most contributing factor to an artificial sound. Computers and machines play things ‘perfectly’ every time, able to hit the same note at a velocity of 127 over and over…but we’re unable to connect with it emotionally.
Therefore, by incorporating subtle dynamic changes, you can create a much more engaging piece of music. This isn’t limited to ‘live’ instruments by any means either.
a) Volume automation
I know it sounds obvious, but it’s surprising how often producers overlook this fundamental tool. In fact, often careful volume automation (and automation in general) is one of the biggest differentials between tracks that impact the listener and tracks that don’t.
After all, it’s easy enough to become lazy and start applying compression far too liberally, especially in electronic music. Ironically, electronic music requires less compression than any other genre, and many producers make the mistake of over use, leading to a bland and fatiguing mix.
There are countless ways that volume automation can be applied to create more interest for the listener. For example, you can increase tension by subtly increasing the volume of a chord sequence throughout a verse. Conversely, you can fade an instrument out to silence to aid in a transition.
You can create fast ‘whips’ with volume curves to accentuate effects like white noise before a drop. You can even apply quick volume changes between the notes in a melody to create tension and a more interesting interplay between notes.
The possibilities really are vast, proving once again that it’s better to experiment with and master the core tools and skills before getting distracted with all manner of third party effects!
You can do a whole lot with volume alone, and it can be very beneficial to limit yourself in this way to develop your skillset.
A fast way to achieving velocity and timing variations is by using the humanise function (or equivalent). While I use this function in Logic, it will be available in most DAWs so make sure you delve into your manual to find out how to take advantage of it.
Humanise does what it says on the tin. It humanises a series of MIDI notes, creating subtle (or dramatic!) dynamic and timing changes that mimic the anomalies of a human performance.
What I love about the humanise function is that it enables me to manipulate velocity randomly within a given range. For example, I can highlight a section of MIDI notes and implement a function that changes the velocity of each individual note to a random level between 80 and 120.
This results in a range of velocity values that, if implemented effectively, can mimic the subtle variations in dynamics of a live player.
This can be applied as subtly or as overtly as you like but, when applied with taste, it can totally transform the sound and feel of your tracks.
For more on MIDI Transform tools, check out this tutorial.
#4 Saturation & Distortion
As mentioned, one of the problems in digital production is that the processing is often simply too ‘perfect’ resulting in an unnatural sound to the human ear. Applying saturation or distortion can add more interest to a sound and can help to create some analog warmth.
a) Tape drive
You can emulate subtle tape drive with plugins like Wave’s Kramer Tape, and there are also freeware plugins available like Softube’s Saturation Knob.
You may have stock plugins built into your DAW, like Overdrive and Exciter in Logic that emulate tape drive and manipulate harmonics.
Not many people know this, but you can actually use some delay tools to emulate tape bias distortion. I know I sound like a broken record (no pun intended) at this point, but subtlety really is key!
This effect is easy to demonstrate using Logic’s built-in Tape Delay plugin, but you can use almost any tape delay emulator to achieve a similar effect.
- Disable ‘Sync’ so that you have full control over the delay time;
- Bring the delay time all the way down to nothing;
- Bring the feedback down to nothing;
- Bring the dry signal down and take the wet signal up to 100%;
- Switch the Tape Head Mode to ‘Diffuse’ to introduce the effect.
You can tweak the Clip Threshold and built-in EQ to customise the saturation, and you also experiment with the tape flutter rate and intensity.
b) Guitar processing
Another useful tip is to check out guitar amp simulators. There are a variety available and many DAWs come with their own stock versions.
Amp Designer and Pedalboard in Logic have some fantastic distortions and you can really dig deep adjusting tone (and even mic placement in Amp Designer).
Just a small amount can add enough grit to make your sound stand out. I would even recommend using distortion as an auxiliary effect on a bus, so that you have even more control over the dry and wet signals.
Plenty of options
Here’s a list of some popular saturation and distortion plugins that I recommend checking out:
- Softube Saturation Knob
- PSP Vintage Warmer
- iZotope Trash
- FabFilter Saturn
- U-he Satin
- Melda Production MSaturatorMB
#5 Modulation Effects
You can use audio effects to transform instruments and vocals beyond recognition, but you can also use them to contribute to a more analog sound.
As mentioned, adding imperfections and nuances throughout your arrangement can accumulate to create a much more engaging sound and a more natural vibe.
In fact, most modulation effects were discovered by accident, so they really can be referred to as imperfections!
As well as harmonic changes through effects like saturation and distortion, you can also modulate sounds by manipulating their time and pitch parameters to emulate many of the traditional techniques used during the times of magnetic tape.
Chorus is essentially the process of duplicating a signal (e.g. a vocal, a guitar etc.) and playing both the original and the copy simultaneously. The timing of the copy is delayed and the pitch is modulated to create a ‘shimmering’ effect where both sounds are perceived as one.
The pitch is normally modulated by an LFO in a similar way to a Flanger, except Chorus uses longer delays and doesn’t employ feedback.
Chorus occurs naturally in many circumstances. For example, think of the term ‘chorus’ used for a choir of singers – a choir singing in unison is essentially one signal being duplicated and each copy having its own pitch modulation. The resulting sound has a much thicker timbre.
You can also hear this in vocal doubling, commonly used by The Beatles and Dave Grohl of Foo Fighters.
Subtly used chorus can add a very nuanced dimension to your instruments, reminiscent of analog tape recordings.
Plugins like Logic’s stock Ensemble are essentially Chorus on steroids, with multiple duplicate signals and LFOs. Check out your DAW’s stock plugins to see if you have any chorus plugins at your disposal.
As mentioned, a flanger is essentially a much faster version of a chorus effect, with one signal delayed by a much smaller and gradually changing amount of time (usually smaller than 20 milliseconds). The resulting sound is a sweeping comb filter where peaks and notches are created in the frequency spectrum.
Adjusting the amount of delay between the signals causes these notches to sweep up or down the frequency spectrum to create the flanging effect.
In addition, part of the output signal is fed back to the input which creates a resonance that exaggerates the intensity of the peaks and troughs in the frequency spectrum (hence the commonly found ‘intensity’ setting in most flanger plugins). This signal can also be inverted in phase, creating yet another variation.
Flanging was discovered by accident. Issues with tape reel machines in the early days could cause signals to drift apart. Particularly when playing back longer tracks during mixdown, the weight of one reel could become heavy, resulting in a discrepancy and the reels going out of sync.
Flanging is a very common effect and you’ll no doubt have access to numerous versions of it in your DAW of choice.
Much like Chorus and Flanging, Phasing is a very popular effect for electric guitar.
Once again, a signal is split and the duplicate is modified with a series of peaks and troughs in the frequency spectrum. These peaks and troughs are then modulated so that they vary over time, and this creates the distinct ‘sweeping’ effect that can be heard.
When the two signals are mixed together, the frequencies that are out of phase cancel each other out which generates the notches that can be heard on the frequency spectrum.
Phasing can be useful for helping certain instruments and effects to stand out from a mix, as the phase cancellation is very distinct. My only warning would be to use it sparingly, creating contrast at specific moments on specific sounds.
You can also try emulators like rotor cabinets for a more old school vibe.
As technology develops and evolves, engineers are becoming smarter about what works and what doesn’t. Companies like UAD are investing a heck of a lot of money into analog emulation with digital tools.
The challenge is recreating the anomalies in analog equipment – the variations, the ‘errors’. It’s these imperfections that bring the sounds to life, and this character and personality is what so many people find so desirable.
It’s simply a matter of time before digital tools can achieve exactly the same as analog. Indeed, many would argue that this has already happened. Soon, our understanding of analog circuitry will mean that there isn’t anything that can’t be emulated and, furthermore, improved.
Do you have your own unique methods of recreating that analog sound in the box? Do you agree or disagree with any of my methods, or know how I could improve them? Get in touch in the comments below and let me know!