Some of you may know these routines and tricks already, but please do read on to refresh your knowledge, perhaps getting your mind to think about new possibilities. For those of you who are completely new to the idea of sends: let’s get into it. This article assumes a little basic knowledge of how to use Renoise as well as some common basic audio mixing terms – so if you get stuck on something, read-up first and feel free to ask a question at the end.

I will firstly look into the routing structure of Renoise to demonstrate the basics of how Send Tracks can be used. I will then look at some common and not-so-common send routings using Renoise’s internal effects. Finally, I will show how sends can be used to form ‘group sends’ to apply common effecting. I’ll drop in some pointers on how to neatly effect and organise sound so that your work lends itself to a better mix.

Basic Use of Send Tracks

The concept of Send Tracks (or ‘sends’ for short) is a mirrored concept from analogue hardware mixers. Hardware mixers have individual channels (or a ‘track’) for each sound – i.e. one channel for the kick, one for the bass, and so on. However, mixers usually have a limited amount of effecting control on each channel, usually just controls like volume, pan, and EQ. If the individual sounds need further effecting then the audio has to be sent from the channel output to external hardware effect units such as tape echo or spring reverb. The sound is altered with effects, then results need to be reconnected to the mixer for mixing and further balancing. Thus, mixers have additional channels labelled as ‘effect return channels’ or ‘receive channels’, so that the mix engineer could further adjust the volume, pan and EQ of the effected sound. The overall process can be referred to as routing.

When digital audio came about, programs copied this concept in one way or another. Usually it is just an additional channel or track that you can send audio to for further effecting.

Renoise does this by using Send Tracks, and a routing tool called the #Send Device. If these things seem totally unfamiliar to you then you need to first learn the fundamental facts from the Renoise online manual. See here for the basic Send Track explanation, then the Send Track in the Mixer View explanation, and finally read the #Send Device explanation. That flow diagram with the green lines is a particularly useful one to come back to in understanding how Renoise organises sending:

Let’s quickly set up a send routing for ourselves. Open up one of your XRNS files, or alternatively one of the Renoise demo songs. In either the Pattern Editor View or the Mixer View click on the ‘Mst’ track (the master channel) and press Ctrl/Command+T. This will add a Send Track right next to the master channel. Take note of the Send Track label, it will be something like S01 – you can rename it as you wish, but you need to remember the label when you are routing later.

Next, select one of the active tracks in your song, something obvious like some drums or a prominent melody. On the bottom left of Renoise open the Track DSPs tab and scroll down the list and double-click the #Send Device from the #Routing section. This will insert the #Send Device as the last item on the track’s DSP effect routing.

By default the device will be set to Mute Source, the send amount at 0dB (full), and the Receiver usually defaults to the left-most Send Track (e.g. S01). So, as per the previous flow diagram, these initial settings means we are sending all the audio of your selected track to Send Track S01. Because you haven’t set up any additional effects, you won’t hear any change to the sound, yet.

Now click on the Send Track you have set up (right of the Mst track remember). We’ll do a quick proof of concept to show the Send Track in use. From the Track DSPs menu insert the mpReverb effect on the Send Track. First alteration is to set the Dry Mix parameter to 0%, and the Wet Mix parameter to 100%. This will ensure that the effect is outputting nothing but reverberated sound and has no original sound mixed in. It’s all wet.

You will hear that your original track sound has become entirely reverberated! Score 10,000 points! *bling!*

Thus far all the effecting has been in-line: but now we will alter the routing so it becomes parallel effecting. Parallel effecting will show the full routing possibility of Renoise’s Send Tracks. So, go back to your original track and focus on the #Send Device. Switch the send mode to from Mute Source to Keep Source. Immediately you will hear the original signal come through the mix dry, at the same time as the reverberated sound. Furthermore, you can control the amount of reverb level directly from the #Send Device by adjusting the Amount parameter. For example, you might find that an Amount of -18dB is just enough reverberation relative to the original sound. Adjust to taste.

In summary, you can now see and hear a parallel effect routing, where a ‘wet’ reverb sound is alongside a ‘dry’ input sound. This illustrates the basic possibilities of using Send Tracks in Renoise. Congratulations: you are now an initiate into the art of sending.

In addition, one of the major reasons for using a Send Track is that we can send more than one sound to it. For example, you want four different tracks with four different amounts of reverb echo added to each; then it is as easy as setting up four different #Send Devices, one on each track with individual Amount adjustments. This is a very neat and efficient way of mixing, and can add a unified feel to your mix due to using common effecting across your tracks. Why have four reverb effects when you can just have one? Use a Send Track. This conserves CPU usage as well.

But, we don’t need to limit ourselves to a simple reverb routing: in fact, the creative possibilities with send routing is limitless and exciting! Let’s explore:

Common Wet Effect Sends

The two most common send routings are for purposes of reverberation or echo. These are ‘wet’ parallel routings much like the example provided above. Let’s look at these common routings closely and see how we can make them optimal.

Continuing with the same reverb routing as above, now focus on the Send Track. Tweak the mpReverb settings to tease out a different sound: Duration of +6000ms; Predelay of 0ms; Low Cut open to 4999hz; Low Gain reduced below -12dB (to filter out low muddiness); Colour Off; Width set to Normal; Pan 7 L (a little to the left); Wet 100% and Dry 0%. Now the reverb is longer and a little more airy feeling.

Have you ever noticed that digital reverb (as opposed to real life acoustic reverberation) sounds lifeless and unconvincing? This problem is especially noticeable with input sounds that are sonically repetitive (like simple drum samples). Well, this problem can be addressed in part by some modulation before the reverberation. To do this, insert the Renoise Chorus DSP into the effect chain, and drag it so it is before the mpReverb effect. Change the Chorus parameters so that it becomes a simple pitch modulator like so: Dry/Wet to 100% (so it is all wet); Rate slowish around 0.4hz; Depth at 40%; Delay at 0ms; Phase to mono; and the rest as is. The Depth parameter will control the amount of pitch modulation and will cause the reverb echoes to have a more liquidised, or organic sound. Colourful reverb!

Next, let’s tidy up the reverberated tone. Sometimes a Low Pass (LP) filter can help focus the sound to the mid range frequencies. Add a Renoise Filter3 DSP to the effect chain after mpReverb, select the 24dB Moog curve and drop the cuttoff frequency to around 4700hz or lower to taste. Another common tone alteration is to EQ some of the hazy excess in the mids. Open up EQ5 on the effect chain after mpReverb. Select one of the mid frequencies and set it to 840hz (or thereabouts), set the Q narrow to about 1.20 and reduce around -4dB. Another common reduction point is 2khz but a little less with around a -1-2dB reduction. Or whatever settings to taste, depending on how the input sound is behaving through the reverberation.

Once you have tweaked the parameters to carve out a superb sounding reverberation you can go back to the #Send Device on the original track and re-adjust the send Amount. Hopefully you will find you have a lovely sophisticated reverb effect using entirely native Renoise effects. I’ve made up an effect chain you can download here and put on your own Send Track that shows this idea in action.

What about some echo? What about some echo?

Send echo (or delay) uses the same approach as above. We want completely wet echo in the Send Track, with possible pre-modulation and tone filtering. Do this first and don’t forget it: enable the Mute Src. button that prevents any of the dry signal to sound in the Send Track. If you miss that button then you’ll get the dry signal messed into the mix. In this example I’m creating a mono echo sound by using the Line Sync, setting both sides to 6 steps. I love epic echoes, so I’m putting both feedbacks to 72%, and everything else can be left as is.

As before, use the Chorus before the delay to cause a subtle or obvious ‘tape wow’ pitch modulation effect. Then you can focus on filtering and EQing the echo tone using LP filters and High Pass (HP) filters. Always go back to your original track and adjust the #Send Device Amount to get the mixing level of the echo right. Finally, you can go through all the tracks and your mix, and see what might sound nice sent to the echo Send Track. I’ve made up an example effect chain you can use on a Send Track to use as a filtered send echo, downloadable here.

Width Effect Sends

Echo doesn’t have to be long and epic. Two differing length short delays, panned left and right are commonly used to improve the stereo image of sounds. Here’s how to do it. Make up a new Send Track with a Renoise Delay DSP with the Mute Src. button on. Set the L Delay to 7ms and the R Delay to 11ms (these can be tweaked to tune in better with your source sounds). Set the feedbacks to 0% so there is no unwanted resonance caused by the short delay. Now go to the L/R Output Pan sliders and ‘swap the channels’ by panning the Right all the way left, and Left all the way right. This will mean that stereo sounds will echo off on the opposite side, which is brilliant for creating additive width.

It’s generally good practice (although not always all the time) to keep low frequencies out of the sides of the stereo image. In plain-speak, this means little or no bass out on the left or right: just in the middle. Because we’ve used the Delay to throw echoes of the sound out left and right, then we can tidy up the low frequencies using a HP filter. For this task I tend to like the HP 4n Butterworth filter set at 777hz or above. Additionally sometimes a LP Filter also works on the wide echoes to roll off detail above 13khz or lower. It depends on what part of the sound you’d like to focus in the width. And as always, go back to your #Send Device and adjust the send Amount until it is just so.

You can use the Chorus to achieve a similar width effect in using a Send Track, with the added bonus that you’ll get some pitch modulation as well. Replace the above Delay with the Chorus effect and set the Dry/Wet parameter to 100%. Tweak the settings to taste: the key attribute that makes the sound ‘wide’ is the Phase parameter. Again, tweak the filters to focus the sound and prevent wide mud. Used subtlety or wildly, a lot of pleasant sounds can be explored here. Which leads us too…

Not-So-Common Send Effect Madness

The Chorus becomes a fairly amazing tool once it is liberated to a Send Track. Have you ever noticed if you’re using the Chorus in-line on an effect chain that the Dry/Wet slider reveals a weaker, darker sound the closer it is to 50%? Try it out. That weaker feeling is caused by a cross-fader algorithm that does not maintain equal power throughout the range. This bugged me initially, as I tend to think of chorus as an ‘additive’ effect: something you add to colour the sound without loosing the volume and presence of the original sound. No big problem though, because we can use the Chorus in a Send Track and use the #Send Device to control the amount of ‘additional colour’ without loosing the feel of the original sound.

This is where the Chorus can be used as a traditional ‘liquidiser’. In this case it is best to adjust the Chorus Phase to 0 so that stereo image remains unchanged. Of course you don’t have to go for traditional: you could go wild with tweaking all those parameters to get a truly mental sound. Go for outrageous amounts of Depth, Feedback and Delay and see where it takes you.

Any why limit ourselves to just the Chorus? Try out the Flanger and Phasers effects as well. These tend to work well in combination with the Chorus’s Delay value use to provide a little separation from the source material. Just a little. And don’t forget third party plugin effects: there are numerous ‘wet’ style effects out there. These include all sorts of creative modulators, spacial effects and glitchy style buffer-divisor effects.

You can really take the sounds to outer-space with creative use of the *Mapping and *Modulation devices (such as the LFO and Signal Follower). Complex modulations and combinations of effects will yield exciting results. You can use this complexity to eek out organic realism (such as analogue variability) or out-of-this-world sonic insanity. Go forth and creatively modulate!

Finally, don’t forget that you can alter where the #Send Device is positioned in the original effect chain. This is another avenue for exploring creative results. An example you might have heard before in mixes involves a standard parallel reverb routing. Try setting the #Send Device to Keep Source and placing DSP effects after the #Send Device. Here you can use a Gainer and automate a fade (in or out) that will serve to bring the dry sound in and out of focus while the wet is continuous in the background. And why stop with a simple Gainer? You can put anything you have available there and effect the dry sound after the send.

Group Sends

The other major way you can use #Send Tracks in Renoise is to do what some applications call ‘grouping’ or ‘bussing’. This is where the sound of many tracks are sent to a common Send Track for group processing. This technique is usually an in-line technique, meaning that all the #Send Devices are set to Mute Source. Obviously, you can send as many tracks as you want into one Send Track, as well as send other Send Tracks into a common track for some complex summing.

The most common application for this technique is to send all the percussion sounds into one ‘grouped’ Send Track. Typically this is where compression, saturation and tone modifying is used to alter the percussion sound in a unified way (sometimes called ‘bus compression’ in analogue mixing).

One trick I like to use on lofi drum sounds is to use the Chorus as a gentle pitch modulator at the start of the group Send Track. This ideas is discussed in this thread, and the general aim is to cause a slight ‘wow’ tape wobble effect that make the drums feel more organic. It can really save your ears from some sonic fatigue.

Once you have a group Send Track established you can send the output of the group  to another Send Track to add a little reverb or echo. Or to any other innovative Send Track you may have made up.

Group sends are a fantastic opportunity to get out the distortion, dynamics tools and filtering effects to make up some unified textures and tones. The sky is the limit with what you can do, so there’s no point in me listing all the possibilities here – it is up to your creativity and what you’re aiming to do.

In Sum

By now I hope you are ready to become a Send Track ninja, and apply the above to your own work. So what are you waiting for? Go back to your mixes and get creative with Send Tracks! Ideas and questions are most welcome in the comments section.

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.

Aim For Reductive

Mixing your music in Renoise can easily turn into a quagmire of EQ settings for every track, and often this can be imprecise and messy. EQ used in an unsure way and often in an additive way ends up introducing more noise than needed, especially when using cheap band-EQs. We’ll discuss EQ in a later article because it’s better seldom used and in place we’ll be using Filter2. A better sensitivity to using the filters will eventually lead to better usage of a quality EQ. It will get you in the good habit of “reductive mixing”.

In mastering a good principle to hold is: “the quality of the mastering is largely determined by the quality of the mix”. So if you have a great mix, then little work will need to be done in mastering. Getting a good mix can be aided by an understanding of the limitations of the digital format, which is the core of your audio-canvass in Renoise. What restrictions you might ask?

Assuming you are working at a digital sample rate of 44,100hz (slices of data per second) then we can already see what the upper limitations are: the highest frequency sine wave (or sound that uses both positive and negative parts of the data) can be 22050hz – because one data slice is needed for the positive vibration and one sample is needed for the negative vibration. I hear you saying, “great” because most humans can’t hear notes much above 18000hz. But, and this is a big but, is a sine wave really going to be a smooth curve at 22050hz or near to that at 44100hz? The higher the frequency the more the resolution of that frequency is lost. The sounds no longer have their analogue smoothness. So, the higher end of the frequency spectrum in digital is usually considered as ‘noise’ or ‘harmonic color’ – no real musical notes are up that high. Most natural sounds and synthetic sounds have content up in that region but a lot sounds differ in terms of the purity or harshness of that content. Because the quality of that noise is being crushed by the sampling restrictions sometimes those upper frequencies are not needed or entirely not desirable.

What about the low end of the frequency spectrum? Here the problem is not frequency but amplitude. Sub-low frequencies below 40hz, because of the Flectcher-Munson curve, take more amplitude to be expressed. We often do not need those frequencies in music because they are mostly rumble sounds: the kick and bass sit higher than 40-50hz. Why would high amplitude rumble be an issue for the mix? When it comes time to master the track, specifically in gently boosting the levels, large amplitude rumble will quash out higher definition sounds because of the lack of headroom in digital audio. Also, excess rumble will make it hard to discern the clarity of the desirable bass elements – hence the term ‘muddy’.

So what is sonically desirable and balanced? What is the sonic model we are aiming for? Here a little listening homework will be necessary! Assuming you have reliable monitors to listen to music on, go get your hands on some recent recordings of classical music. Yes I know, classical might not be your thing, but you’ll find no more reliable reference material in any other popular genre. Most classical music has been recorded for optimum fidelity in mind, so that it near enough simulates the experience of actually being there at the gig, hearing all the acoustic instruments in pure form. You’ll find that the audio has a clear and deep quality to it that we can use as a sonic model for mixing. Try to find something that has a range of loud and soft in it, like easy stuff such as Beethoven’s 9th Symphony or something more special like Arvo Pärt’s Tabla Rasa. Listen to the balance of the frequencies: the highs are never too intense, the bass is present but not muddy. Study the sonic expression of the sound, and then use this benchmark to apply a sonic model onto your own mixing. To get anywhere near the clean expression of sound you’ll have some filtering to do.

We can use Filter2 to get rid of unnecessary frequencies to get your individual tracks as clear as each instrument in the classical model. It’s a very powerful and underused tool that I have on almost every track of every one of my Renoise songs. Every sound, in every song, within every genre, will all have different needs and the best way to find what works is to get your hands dirty with trying different settings.

But a few explanations of the different settings in Filter2 first. I’ll be using some shorthand terms: “Cut” (cutoff) refers to the frequency cut-off of Filter2, and “res” (reso/Q) refers to the resonance or amount of feedback Filter2 is using. Next, choosing an filter with the right ‘slope’ is important. The slope refers to how sharp the cut-off is, e.g. -12dB refers to the volume of the sound dropping away from the cut at a slope of 12dB per frequency octave. So, a slope of -48dB is steeper than -12db. The Moog filter has a strong immediate cut and thus has a slight resonance to it suitable for automated filter sweeps. Also, khz is a fancy way of saying 1000′s of hz.

What follows is some of my common usages of filter2 for making a slick mix:

LP Filtering

LP filters are useful in knocking out unwanted high frequencies of each track. Here are some common examples:

• Temper distortion: a Moog or -12dB cut of anywhere between 4khz to to 12khz. This makes the distortion sound smoother and warm. Usually the larger the amount of distortion the lower the cut. Sometimes just a small amount of res to brighten the cut point.
• Reducing the hiss aspect of old samples – when you simply have no alternative. A Moog cut around 12khz-16khz usually works.
• Harsh high-hat and cymbal sounds can be fixed with a Moog cut around 14khz to 16hkz, sometimes with 2-3% res to re-compensate for brightness. Alternatively you can use a smoother -12dB roll off depending on the characteristic ‘sizzle’ of the sound.
• Some pad/string sounds really benefit if you filter out the highs, especially saw-tooth Juno-like pads. Here you can go as low as you like, but also use a small amount of resonance relative to how ‘distant’ you want the pad to be.
• A deep cut on a kick drum, sometimes all the way down to 100hz can have a nice effect for a pulsating sub-kick heartbeat, suitable for all sorts of genres.
• Any other sound that doesn’t need the upper frequencies expressed – it’s almost always better to cut the tops than to have a really busy noisy upper range.

HP Filtering

HP filters are useful in knocking out unwanted low frequencies of each track. It is ideal to have a tuned sub-woofer with your speakers so that you can get accurate monitoring all the way down. Here are some common examples:

• Many kick drums benefit from having a -12dB HP filter with a cut around 90-110hz. This keeps the kick punchy but rolls most of the subsonic amplitude.
• Taming the bass – A -12dB HP filter with a cut around 120-200hz depending on the character and key of the bass sound. A small amount of res (less than 6%) will boost the low end punch without over doing the subs.
• Keeping vocals behaved – A Moog HP at 80hz to take out any dangerous plosive amplitude bursts. Great for any acoustic recordings.
• Handy on counter-percussion/loops – use any of the HP filters above 300hz to take out the lows so they don’t interfere with your main kick drum.
• Any other high melodic instruments will benefit from clearer expression without their low component – experiment and find points that are suitable for your sounds.

Using Two Filters

Quiet often it becomes necessary to use both of the above techniques in conjunction to really focus down onto the necessary sound. Yes, you could use a Band Pass filter, but I prefer using two Filter2 plugs so that on each I can have the freedom of using different slopes and resolutions. Here are some common examples:

• Reverb and Delay decay – Cut the lows below 600hz or higher and cut the highs with a Moog HP 2k or higher. This will help to get a more authentic echo and one more reminiscent of the cool analogue dub delays of days gone by. Because of the narrow frequency range the sound will become more distant (therefore magical).
• Kicks and Basses – as discussed before in HP filtering, but used in conjunction with LP filtering at any desired setting. Taking out all the high sounds in the low instruments will free up the canvass in the mids.
• Narrow the pads and ambience – just to the mid-range with resonance on the LP filter. Again this will improve the illusion of distance.

Ready, Set: Filter!

Of course, there are probably millions of more examples that could be stated on how to use filters to shape the sonics. Because of the wide variety of genres most of these examples will be fairly academic, and are best found through experimentation. The general principle holds: reduce bass for control of sub-rumble and mix-amplitude, reduce highs to lower digitalized noise. Once you have tidied up your mix in this fashion you’ll reap the rewards in the mastering process – and – your precious composition will be all the more enjoyable for an audience.

This is fairly basic stuff and pretty obvious to most, but since the topic came up a couple of times in the IRC channel, I have made this basic tutorial with pictures to visualize the filtering process.

I have used the free effects from GVST. You can get them here: http://www.gvst.co.uk

Whatever is explained here also applies to the internal filters of Renoise.

First let me start with a short overview of the most useful filters. After that I will provide you with some tips on how to use them. All filters have in common that they don’t introduce a hard cut. That means when I use a low pass filter at 2kHz, there is still some activity above that frequency. That activity can be accounted to a smooth transition that filter have, which is called the slope. If the technical stuff is interesting for you, you might check the following links.

• http://en.wikipedia.org/wiki/Low-pass_filter
• http://en.wikipedia.org/wiki/High-pass_filter
• http://en.wikipedia.org/wiki/Band-pass_filter

Have a look at the following pictures. The red parts in the spectrum view are removed by the filter. Moving the knob in the VST will relocate the red parts accordingly. The blue parts are passed through to the output.

The Low-Pass Filter

Just as the name says, a low-pass filter will remove high frequencies.

The High-Pass Filter

A high-pass filter is the opposite of the low-pass filter and removes low frequencies.

The Band-Pass Filter

A band-pass filter removes low and high frequencies. A band-pass filter can also be simulated by chaining low- and high-pass filters.

Tips

If you have a song with lots of instruments playing, there is a high chance your mix is overcrowded. Many instruments will overlap each others’ frequency regions. Filters are really useful in such situations to make room in the song and get a more transparent output.

A pad, for example, usually doesn’t need much low content, so you can high-pass filter it. When soloing a pad with a high-pass filter it can sound thin, but when played together with the rest of your song it will probably sit well in the mix.

Kicks and basslines are perfect subjects for low-pass filters. Often the bass department of a song sounds muddy. You can decide whether the kick or the bassline should be filtered out. Let’s say you want to duck the bassline on every kick, to get that popular pumping effect:

1. use a low-pass filter on the kick
2. put a band-pass filter on the bassline
3. adjust the low cut of the band-pass filter to make room for the kick

A lot of people are after some nice kicks for their songs. One method is layering a couple of them, then use filters on the layers to extract the interesting parts for your new kick.

Maybe you have found some useful information here. The filter settings ofcourse depend on your taste and on the sounds you use, so experiment a bit. Feel free to comment and add your own tips.