This article deals with how to use speakers to get a great final mix for your song. And you may be surprised to see just how low budget you can pull this off with. Regarding technicalities with software, it is ideal that your read my other recent In:Depth article on avoiding clipping. Especially read in regard to setting playback loudness levels. The approach addressed below can be applied for mixing in Renoise, but also in any other DAW or mixing projects.

The need for a final mix

I like to think of a good final mix as equal parts ‘creative art’ and ’scientific balance’. I’m assuming that you have the ‘creative art’ aspect of the deal well established. This part is your compositional ideas, your choice of sounds, instrumentation, transitions, effects and sonic shape of your music. These are the elements that give your song unique character. Think of them as indisputable, and as the representatives of your intentions as an artist.

The second part of what qualifies for a good final mix is what was once strictly the domain of mix engineers: the precise and refined shaping of the sound in a near-scientific balance and flow. Any mix engineer worth their fee would honour the ‘creative art’ of the mix. They use tools like volume, pan, EQ, and compression to focus and clearly present the music. Many engineers weren’t actually creative musicians, but more like technical craftspeople who would approach a mix analytically with great skill using the best equipment possible. These technicians eventually had it very lucky, because their great sounding equipment and production procedure made it more and more effortless to get a very good result. Given this, why should it be so hard for you?

These days you can attempt it all yourself using Renoise on a computer in your bedroom. Where a lot of home producers run into problems is not being able to make the transition from ‘creative art’ to ’scientific balance’, or even knowing they might need to make that transition. These two parts of production require radically different mind-sets and skills; but of course no one tells you that when you start making your own music. This is why it is ideal for home producers to break up the production into these two distinct parts: first the art, then once the art is complete the final mix sessions can begin. Even better is to distance that final mix session work well apart from the creative stage, so there is no temptation to go back into the song and add indecisively. Separate production stages help the mind focus on different processes and issues. It gives you a better chance at making that final mix one that has had some dedicated and well thought-out work done on it.

The better a final mix is the easier and more natural the mastering of the music. Mastering is a whole other technical area, and I will discuss this in future articles. For now, it is important to understand that good final mix prepares the sound for a good mastering. The mix has headroom and no clipping. It will lack excess and unnatural emphasis in any frequency.

All well and good, but what about all that tweaking and fine tuning to get things clear and balanced? You might feel a little daunted and have no idea where to start with all these volume settings, panning settings, EQs, and compressors. For example, just look at EQ10 in Renoise: there are 30 different parameters to tweak and hope that what you’re hearing is doing the sound some benefit. So many parameters, what to do with them all? To help guide us, we need to look at a phenomenon called ‘translation’.

Defining translation

Some of you might know the science fiction series The Hitchhiker’s Guide To The Galaxy. In this story there is a thing called a Babel Fish, which is a small fish people put in their ear. The fish amazingly translates any language in the galaxy into whatever language the fish’s person understands. Now, wouldn’t it be neat if there were such a thing that could do this for our mixes? Well, last time I checked there isn’t Babel Fish you can buy that automatically translates sloppy mixes into audio-gold – but the good news is there is a way of doing this yourself to get best results on any speakers.

This achievement is called translation. Translation is where the mix is clearly balanced and organised so that it sounds great and understandable on any speaker system. It is like the ‘ideal packaging’ the delivers the product (being the creative ideas) in the cleanest and most efficient way possible. Translation is genre agnostic. Crazy noise music can be expressed just as clearly and beautifully as the softest whisper of acoustic recording or programming. It never holds the intention of sounds back, but rather presents them in a way that is unmasked, organic and sonically pleasant. Mixes that have achieved translation tend to also have a longevity to them – you can listen to them over and over again and your ears do not tire from the sound.

Translation can be mostly achieved at the final mix stage. The more you achieve in the mix, the less you have to squeeze and contort in the mastering. So now, you know you’re using your Renoise mixing effects to try and tweak things to get this phenomenon called translation. But, before you even begin tweaking, you need to consider how you are using the very things that reproduce the sound you are trying to shape. Speaker monitors.

One set of monitors: the pitfalls

All sets of speakers have a frequency response. This is a measure of how accurate the speakers represent all the audible frequencies: from subs, to bass, to mids, and to highs. If the speakers have been tested scientifically, then the response of the speakers can be graphed like so:

Some speakers have a more ‘flat’ looking response than others. The flatter the response, the better the speakers are for accurate monitoring of your sound. Generally speaking, small consumer and hi-fi speakers don’t have the flattest response in the world; and sometimes they even over-emphasise certain areas to make them more marketable. Professional studio monitors are usually made of higher grade material and technology, and have a flatter response. But, there are absolutely no set of speakers out there that have a dead flat, 100% perfectly accurate response. They don’t exist! So even if you’re buying a $20,000 set of monitors, you’re still getting an imperfect representation of the sound.

Additionally, different sets of monitors have different ‘dynamic response’. This is the measure of how accurately they represent the dynamics of the sound (the transitions between loud and quiet). In my experience, some speakers sound ’shallow’, while others are more ‘expressive’.

Add also to the equation the aspect of amplifier colour. Any signal that passes through circuitry that has op-amps will be coloured. There are solid-state transistor colours, as there are a large variety of tube amp colours. And, there are different amp designs that produce different colours. A whole rainbow of confusion!

To make matters even more complex every single room in sounds different. This is the realm of acoustics: the frequency response of a room. Frequencies bounce around in room producing resonances and cancellations, which in turn distort the perception of the flatness of the sound. Mid and high frequency reflections also make the room more lively, making it difficult to understand the dynamics. Of course, professional studios are treated to manage acoustics, and there are many schools of thought on how to best do this (more on this in another article). But even with the best treatment in the world, the room will still be imperfectly unbalanced in some regard.

Factor in all of the above, and then make the consideration that some engineers and producers insist on on using only one set of monitors! Can you believe it? Sure, there’s some weight to the point that ‘knowing your gear well’ helps you work out a good sound, but why risk it? Unless you know your monitors utterly and absolutely, it is going to be difficult for you make a good final mix happen, let alone achieving translation.

The pitfalls are as follows. Where your monitors, gear and acoustics under-represent a frequency (e.g. a bass sounds less than it actually is) you will tend to over-emphasise that part in your mix. The reverse is true: if your gear is over-stating a frequency then you will naturally hack back at it and scoop it too much out of your mix. The more you become reliant on a deficiency the more you develop bad habits that ruin mixes. I think it is safe to wager that 99.99% of us are working with gear and environments that aren’t 99.99% flat – so why would we limit ourselves to just one set of monitors who steer us into one way of working?

The trick is use more than one set of monitors. Routinely.

Triangulation

I’m not sure if this term has been coined elsewhere, but I call this method of using more than one set of monitors ‘triangulation’. A triangle has three sides, so I use no less than three sets of monitors to balance out and cancel out any biases I have from one set. The triangle analogy also works well for the ideal listening position for near-field monitors: at equal hight to your ears in a triangle pointing the speakers at your head. So whenever I am doing a final mix and trying to achieve translation, I do so by checking the mix through triangulation.

I have some personal rules for triangulation. One set of monitors must be large-diaphragm enclosed headphones, preferably of reputable quality. Another set must be some reliable, and flat as possible near-field monitors, preferably on stands in an acoustically treated room of a largish size. A third set of loud hi-fi speaker (or near-fields if you have them) should be used in a different room (again a large room is better at volume). Optional extras include: a second set of near-fields in the studio; laptop speakers; earbuds; and car stereo. However, these last few are not reliable for editing work, only ‘end checking’.

Here is a breakdown of my own final mix routine. I do most of my mixing work on my headphones, which currently are AKG K240 mkII. I do a lot of that mixing at fairly low volume in a quiet environment. This is so I preserve my ears, and efficiently do a lot of obvious or general editing. The ‘cans’ are really good for setting dynamics. Good headphones are also ideal for really tightening up the placement and expression of low frequencies, as you’re getting no acoustic interference. I do a lot of general listening to CDs of all genres on my headphones, so I really know what a good low end frequency image feels like. Incidentally, I really recommend headphones for composition, as they are great for really getting immersed in the sonic universe of your song and being creative with that.

Once I’m happy with the headphones I move the mix onto my Yamaha HS80Ms, which are on sand filled stands in my acoustically treated project studio. The pandas (my nickname for the monitors) have 8” bass drivers, so they give a very open representation of the low end that I like and tend not to feel from 6” drivers. In the tradition of the Yamaha’s NS10s, these monitors are bright and full on with the upper mids, so they are great for detailed EQ work in tempering excess in those areas. The mix is checked at different volumes to get an accurate sense of dynamics and the inertia of frequency ranges. I also sum the mix to mono to check all the volume placements, and then back to stereo to fine tune the pan positions or other spatial effects.

Then I move onto a second set of near-fields I have, which are in the same room, on foam and upon my desk. These are my old custom-built monitors running on a budget Class AB 50W amp. They only have 4” bass drivers, and man do the high frequencies rip and squeal! They are just about downright painful to listen to, but are very handy for double checking excess in top frequencies as well as how well kicks and basses are articulated on gutless speakers. A check on these doesn’t take too long.

Finally I’ll take the mix out to our living room, which is a long large room, with an odd shape and no acoustic treatment. There I have a Class A Technics SU-700 amp hooked up to two JVC SK-700II hi-fi speakers. I play the mix loud and annoy the neighbours! These speakers have a very soft, musical and dynamic quality to them. They are a joy to listen to music on for hours and hours. I use them as a final check for translation as well as to see if I got the brightness of the mix present enough. If I hear anything amiss, I go back to the studio with fresh ideas on how to tighten things up.

Using this triangulation, I can achieve final mix translation. I can take a mix-down and play it anywhere knowing it will work. Then I can graduate the song to the mastering stage. And you know what? I use exactly the same triangulation method in mastering as well!

You don’t have to go out and buy all the same gear I have and use it in the same way. The main point is that you use at least the three of the most reliable monitoring methods you have available to you. By all means, if you have a lazy $20,000 to go spend on an amazing set of monitors, go right ahead! But without checking your mix systematically on two other sets of speakers, then you can only be guessing if you’ve got translation or not. I think you will find that you don’t need an outrageous budget to set this situation up.

Final checks

Using triangulation in your monitoring procedure will help you begin the task of accurately using the Renoise mixing tools such as volume, panning, EQ and compression. The specific use of all these tools could produce a whole article for each technique and each situation. No point in beginning to learn all those tools without a reliable monitoring method.

You’ll know when you have translation: the mix will have an open and clear feel. It will feel like it just ‘works’. A final mix doesn’t have the full punch, width and brightness a mastered track can have – but it will feel absolutely ’safe’ and well presented. All the averaging you will have done across different monitors will reveal the gold.

Go forth and experiment. Find an adaptation of these ideas for your own situation. Then, hopefully you’ll be able to share those mixes with friends and online without fear of unwanted sonic surprises.

Before getting deep into this article it is best that you make sure you’re fully familiar with Renoise’s DSP audio mixing system, which can be found here in the Renoise Manual. If you’re unfamiliar with any of the Renoise terminology used here it is always good to go back to the tutorials to brush up on some technical knowledge of your favourite music software. This article also uses a lot of common audio production terms – so if in doubt look the word up. I’ve linked some of the terms so you can follow up on with further reading.

What is clipping and why avoid it?

Clipping is a form of distortion where the amplitude of the waveform is attempting to be so large it runs out of room and pushes against the extreme edges of available sampling space (referred to in digital audio terms as ‘0dB’). Because it has nowhere to go the waveform is flattened out at the peak of the amplitude in a hard line until the amplitude lowers again and resumes its normal curve (see the two waveform diagrams below). Both the hard line and the fact that there is a tight little corner bent into the shape of the waveform contribute to audible distortion. The sharper that bend and the longer that line is straight the harsher the sound of the distortion. If you’d like to know more about the technical background to clipping and distortion, start reading here.

This is a healthy looking waveform. Plenty of headroom here. But look at this one:

This waveform has been clipped. It is the same sound as before but with +12dB added. You can see the edges have become squared.

This of course is sometimes a desirable sound effect, but for the case of making a very clean final mix on your song it is not desirable. Not only can clipping sound ugly, it can prevent a good result in the mastering process later on after the final mix is done. Why is this so? Mastering initially requires the final mix to have at least some ‘headroom’ to work with. The room to move is needed when applying further effecting to the waveform, which is commonly further boosting the amplitude. Headroom is the amount of space left between the loudest peak amplitude (say -4dB peak) to the absolute edge of the digital audio envelope at 0dB. If your mix ‘peaks’ at -4dB, that means you have 4dB of headroom, and probably ample space for mastering to do its job. In case you haven’t figured it out by now all the loudness values in digital audio are expressed in negative dB. -4dB is quieter than 0dB. Look at the following example:

If your own mix has peaked at 0dB and the clipping light has turned on, then you have distortion and no headroom left for mastering. Time to rethink the plan!

Ideal master channel settings

Doing mastering for clients over the years has allowed me to see some pretty varied use of the settings on the master channel, and some songs have been downright wacky! So, I’ve developed settings and a method that helps prevent mix clipping and leave a usable amount of headroom. First, let’s look at the Track DSP strip for the master channel (Mst):

Take a look at the settings in this image. Don’t worry too much if you don’t know what every single thing here means, we are only examining the chosen settings for this task.

Let’s go through each relevant setting: Panning is centred so that nothing is pushed unevenly out to any of the sides. Pre-fader volume is set and fixed to 0dB (very important!). The Width is not used at all because we don’t want to introduce strange short delays to the mix and muck with the stereo image. The post-fader is set and fixed at 0dB (again very important!). DC Filtering is off because it’s better to clean up the low end frequencies selectively in the mix. Soft Clipping is off for two reasons: one is that we don’t want to hide the fact that the mix could be clipping; the other reason is we don’t want any subtle alteration over the entire mix potentially colouring the peaks in an undesirable masking manner. The Auto Gain is off so that Renoise is not altering the post-fader volume behind your back, thus making any potential clipping appear straight away. Overall this approach is very transparent – letting you hear the mix as is without any alteration.

I can hear some of you say “Hold the bus! I don’t set up my mix this way” Well, that’s ok, but if you want to get a very clean mix without clipping this is the best way. It may now become apparent your song sounds too quiet… Well, time to turn up your monitors! This brings up a wider issue about monitoring.

How to set up your monitoring levels

You may have heard of the term ‘bit-perfect’, and if you would like to learn more about bit-depth issues you can delve into the subject here. Without heavily going into the details it is important to adjust your playback volume of your monitors (speaker or headphones) using an analogue fader, not a digital one. This means your soundcard channel volumes (if the card has a software mixer) all have to be set at 0dB. This will prevent bit-level degradation to the sound. Some people’s studios have an amplifier or hardware mixer connected to the soundcard output before it goes the monitors – adjust the playback volume there. Similarly, if your monitors are active and have rear attenuators, you can adjust the playback volume that way. Or, you can do what I’ve done and get a passive analogue volume knob that sits between the soundcard output and my active monitors (see picture below of my big knob!). Whatever method you choose, it will sound better if you’re adjusting the playback volume with an analogue fader – digital adjustment isn’t bit-perfect and can kill representation of dynamics at settings other than 0dB. If all of this sounds a bit too daunting for you then you can just adjust the master output fader on your soundcard, and make a project of later on getting an analogue fader to do that job properly.

The other major point with ramping up your monitor playback volume is that you might use your computer system to play things other than your Renoise mixes – for example: mp3s, DVDs and CDs. These might now sound just way too loud! What to do? Because you’re using the master channel settings I outlined earlier your mixes will sound loud enough, but anything that has had commercial mastering done on it is going to sound way too loud. So, you’ll need to roughly decide on two monitor playback volume settings, each depending on what you’re doing at the time. This is where having an analogue fader is really handy – I put mine at 11 o’clock for listening to CDs, and way up to 2-3 o’clock for working on final mixes in Renoise. Your settings will be relative to what gear you have.

Commercially mastered material has boosted loudness, sometimes excessively boosted (so much so that they’ve given a name to the phenomenon: The Loudness War). I want to stress that a final mix of your music in Renoise is not as loud as commercially mastered audio. Do you remember from before that a final mix needs to avoid clipping and have some headroom? This is why you will need two different settings for your overall playback volume – a louder setting for working on mixes, and a quieter one for playback of commercially mastered audio.

Final mixing – avoiding clipping

So you’ve got this far and set your master channel for clean uninterrupted output, and you’ve set your overall playback monitoring volume level suitable for working on a clean mix with headroom. How do you avoid clipping? Initially this may seem as simple as ‘turning down the volume’ on individual channels – but more it is more likely there are other elements you can fix up with more precision. What are some of the common things that cause mix clipping, and what can you do about it?

Excessive bass frequencies are almost certainly the number one culprit of causing mix amplitudes to be so high they are clipping. There is a whole separate art form of getting a final mix tonally right so there are not any excessive frequencies in any area. We’ll look at aspects of this in future articles. Another related issue is the need for accurate monitoring – having speakers that represent all the frequencies in an honest way (which is yet another topic we’ll look at later). The general point though is lower frequency sounds require more amplitude to express themselves and be heard on an equal level (see the Fletcher-Munson Curve for an explanation). Because of this need for equal expression, bass frequencies can quickly get out of hand and excessive. You may need to give attention to those frequencies by closely EQing sounds like kicks and basses. Reducing low end, especially sub-bass frequencies in sounds where they are not needed will also help clean up the mix and avoid needless excess in amplitude.

Way too much booty!

Snappy large transients of a quick nature can also ’spike’ through your mix and cause unnecessary clipping. What is a transient? Take a look at a the waveform of a snare sound – the transient is the part of the sound where it is the most ’spiked’ – a quick transition of sound having a peaking percussive quality to it. Transients are very common in percussion sounds, but can also be found in snappy sounding instruments like funky guitars or plucked synth sounds. These quick sounds can sit above the rest of the mix sounds, and sometimes so much so they cause clipping. This can be remedied in a number of ways. One way is to apply gentle compression that addresses the excessive part of the transient. Compression using a shorter attack in order to be quick enough to catch the spike will make sound ‘behave’. Another way is to apply very gentle and light digitally emulated saturation on the transient sound, which will reduce the snappy dynamics of the sound. It goes without saying that compression and saturation can be overdone, and that’s something I’ll cover in detail with future articles.

There be some stray transients going on!

Summing up…

In summary: if in doubt reduce something! You may find yourself reducing just about every channel in your mix to get some reasonable headroom happening. Sound too quiet? That’s ok, just turn up your monitors! The more you work according to this method, the more confident you’ll get in making a slick and clean mix without clipping. You’ve got at least 24 bits of data resolution to play with to express all your dynamics, around -144dB worth of dynamic range! That is a lot, so use it! No point in slamming it all up near 0dB risking clipping. Some quieter styles of music don’t even peak over -6dB in their final mix.

I’ve made up an example XRNS file that’s at the final mix stage. To play this file you will need at least Renoise 2.5. It will show you an example of a simple clean mix without clipping, and you can use it as a benchmark to set your playback monitor levels. It is also an example of a neat final mix that is ‘ready’ for mastering. So, it won’t sound ‘done’ because that’s what the mastering achieves.

The example song also shows a little use of the master pre-fader for the purpose of a fade-out (see the last pattern). If you songs need to use a master fade-in or out, then automate the pre-fader on the master channel up to or down from 0dB. That way, for the bulk of your song it remains uninterrupted at 0dB. So make sure that your automated maximum points are precisely set at 0dB.

Whether you’re going to do the mastering yourself later on, or passing the song on to a mastering service – you now have a good idea how to complete a final mix without the major sonic disaster of clipping. The cleaner and better presented the final mix is, the easier and less destructive the mastering will be. Good luck! And if in doubt, share your mix with the Renoise community and you’ll surely receive some great tips and feedback.

Speaking of Indamixx, we are running a challenging music contest. We are giving away one of their Netbooks for the best low-power Renoise song. Go to CreateDigitalMusic for the details. You need to send in your music before 15 October.

The groove settings feature is great, but only applies to the complete song.

Also, the alternative methods of  speed and tempo tricks,  explained on the same page, affect the whole pattern for the rows used.  So, how do we isolate things further?

In the following movie we are going to show you how to generate a simple 50% groove pattern on a single note-column and at the second part of the movie, we will demonstrate the same method to apply a groove pattern to the full track.

Also, this trick does not affect synchronisation problems to external hosts or plugin effects that cannot handle swift timing effects, which is an extra plus benefit as well.

To clarify, the following procedures were used in the movie…

-We pick a delay value of 2 to apply on each second row.
-To apply a 50% groove, everything needs to be sliced by 2, else there won’t be a 50% groove, so we place a delay command on every second row too.
-Then there is the amount of delay we can apply and as we work with individual tracks or notecolumns, we can only use the delay command to achieve this local effect and the delay command is affected by the speedfactor which you have to divide by 2 as well.
-The example song in the movie used speed 6 which is 3 when you divide this number by 2. However…
-Each row starts at the first tick which is position 0(zero) for Renoise and not position 1. Basically a speed to effect command translation table would look like this:

So for that reason, we say speed divided by two minus 1 which means: 6 / 2 = 3 -1 gives you the value of 2 for the delay effect command.

For every command that is based on ticks, this table applies. The higher the speed value, the more ticks in a row you have, the higher the integrity of your effect commands can be.

But that falls outside the scope of this tutorial. (You can read more about that at this location.)

That was for the background information, now go and watch the movie…

Note that you need to manually pause the movie when you see a text balloon as they swiftly pass by during the movie