013 – modulating filters – three of three

Over the last two posts I gave you a run through of filters and equalizers, and their basic uses and best practices. This week I’ll finish up this series by talking about some modulation options to be explored with filters and equalizers. Modulating these processing tools is incredibly valuable in adding dynamics, depth, personality, and interest into your tracks and mixes, and in some cases can make the difference between an ok sound and an awesome sound.

Within most synths, both hardware and digital, there is a filter envelope control which can be set to positive or negative time influence on the sound. When I am creating my own synth patches, I’ll often ignore the amplitude envelope entirely after setting the fastest attack possible, and a decay/sustain to suit the rhythm of the track. The additional motion within my own synth patches is often entirely contained within the filter envelope and sometimes some lfo routed into the filter as well. By adjusting the time parameter from negative to positive on the filter envelope, we can completely change the feeling of the sound from one note trigger to the next. Of course, not all synths include the negative amount parameter, but most do. We can also do this with pre-recorded loops, and live’s built-in auto filter. The envelope there can even be triggered by a sidechain input to create a completely different filter rhythm than you would otherwise get from a standard envelope. These frequency cutoff modulations can be combined with drawn or recorded resonance modulations to add emphasis to the moving cutoff frequency.

As I briefly mentioned above, lfos can also be used to great effect with filters, changing (usually) the cutoff, resonance, or envelope parameter of a filter. Within most soft-synths this is really easy to route together, and if all else fails the lfo plugins available via max4live are incredibly powerful for this sort of modulation. I like using lfos to adjust parameters, but also going into live’s clip modulation window and changing the parameter modulation in some awkward, unlinked loop length. This way I can add a crazy amount of resonance or a weird cutoff jump maybe every eleven bars, or three and third bars; something completely unrelated to the structure of the track and the lfo rate. I usually stick with quantised and synchronised lfo rates as well, so this off-beat modulation can sound particularly powerful, as most of the modulations are locked to the grid.

To finish up the post for this week, let’s also briefly go over parallel equalizing. I’ll hit parallel processing in further depth in a future post, but this is a quick intro in relation to filtering sounds. If you are finding a filter modulation to sound a little too powerful, but it’s still an important element in a sound, you can mix in the modulation as a part of the full sound, and let unprocessed sound through as well. Some filter plugins will give you a dry/wet control, but most will not. Live gives you a quick workaround using effect racks, and I use these a lot when I am processing tracks. Simply create two chains within the rack, one wet and one dry, and adjust volumes to suit the mix you want. The filtered chain can be combined with compression, delay, reverb, or anything else which compliments the sound and the mix of the track overall. Again, I will get to parallel processing in further depth later on, but this is a quick idea as to how to incorporate the technique with filters and equalizers.

Tune back in next Monday for some more production tips, and don’t forget to check out my previous posts for more tips, tricks, and extra knowledge you can annoy your friends with.

012 – eq – two of three

Last week I dove into filters, and this week I will continue on that theme with equalizers. Equalizers seem to be one of the things beginners get hung up on, and one of the processing tools most useful for mixing.

At it’s core, an eq is a combination of various types of filters. We started with set-band eqs, with a programmed curve, resonance, and frequency band, and the only possible adjustment was gain. Since then audio engineers have made much advancement, and the world of digital has allowed us even more capability. The most common eq most people are familiar with in a daw is the parametric eq, in which there are multiple bands which can be set to any filter type, frequency, resonance, and a selection of slopes. We can use these simply as filters, cutting out unwanted frequencies, but we can also boost content we want to emphasize or a combination of techniques.

A question many people have when they start to work with eqs, especially parametric digital equalizers, is what do I boost, what do I cut, and by how much? The answer is, of course, it depends. The first, and most important thing to keep in mind is that eqing should always happen in the mix. This is important to ensure all elements of the track work together and sound awesome together. You could have the best kick sound in the world, but if it doesn’t gel with the rest of the mix, it doesn’t matter. So, with that in mind, I will run through a few tips for using equalizers.

Start by cutting the frequencies you don’t want in the sound, although some people will say you should not do this but instead choose samples without extraneous frequency content you do not want. I will engage with that argument in a later post.

After cutting, you can start to gently attenuate or boost parts of the sound you want to diminish or emphasize. Depending on the sound and what your ears tell you about how the sound works in the mix, you can use shelf filters, notch, or a band filter, or any appropriate combination. When boosting, be aware of how much gain you are adding to the sound. It is entirely possible you will start to clip the sound by boosting the volume over 0dB, and that will sound bad. If you still need to boost by so much, use the output gain on the filter, or use a utility after, to match the input and output gain. This is important, not only to prevent clipping but also to allow your ears and brain to make the right mixing choice. A louder sound will almost always sound better than a quieter sound, thus matching input and output volume is crucial in making accurate decisions about how an effect sounds and if it is necessary.

When using both an equalizer and a compressor on a track, it is best practice to locate the compressor after the eq. This is to ensure you are only compressing the heard sound, and the compressor is not attenuating the gain based on a louder frequency band which the equalizer is removing.

I will conclude this post by mentioning dynamic equalizers. Examples of such are the free tdr nova, and the excellent waves f6. A dynamic equalizer combines the filters of a parametric eq with the active gain attentuation of a compressor. Depending on what problem you are trying to solve, or what type of sound you wish to achieve, dynamic eqs can be incredibly useful. It allows you to boost or attenuate a frequency band, and then apply a compressor on just that band. The compressors range can often be inverted as well, to boost a frequency but only when it crosses a set gain threshold. These compressors can also be sidechained, to allow you to move one sound out of the way of another in a very precise manner. I wouldn’t necessarily use these on every track or even each project, but they are a useful tool to have and have some knowledge of how to use.

Check back in next week for some tips on creative sound design effects using filters and equalizers.

011 – filters – one of three

In the next few blog posts, I will briefly explain and offer some use case examples for a selection of tools and techniques which I find to be rather misunderstood. These tools will also combine together nicely, so in a few weeks’ time, you’ll have a better understanding of where to go with some of your creative and mixing decisions.

This week we will begin with filters, and next week I’ll finish on that theme by running through equalisers.

Filters are incredibly simple at first, and equally, incredibly important. As with all effects, they can be used both creatively and technically; correct use will not only clean up your mixes but can also add much more depth and interest. Filters usually come in four primary types and a few other flavours which are combinations of the four basic ones. A low pass filter is used to remove higher frequency content, thus the name; the lows pass. Conversely, there is also the high pass, which cuts out low frequencies. The next two can be viewed as combinations already, a notch filter allows all frequencies to pass except for a specified band; the notch. A band pass is the opposite, and will only allow a specified frequency band through. Your standard filter will have a few basic settings, filter type, cutoff frequency, resonance, and slope. More controls can be added when the filter is more advanced or designed for different purposes.

How do I decide which filter to use and where? As with any effect, the first thing to ask is, what would I like to do? It seems completely obvious, but it is completely common for many people to throw on a filter or eq, and a compressor, onto every channel – either because they can or someone told them that was a good idea. Some tracks and styles may need all that, sure, but most don’t. So start by deciding what you want to do, and if a filter helps you get there, we can move on. If it’s a simple cut of offending or muddying frequencies, we can simply make that cut, but we don’t have to stop there. Filters have always been used as creative tools as well as precise technical tools. For my ears, this means using specific analogue-modelled filters to achieve a certain colour or distortion to the sound. Clean digital filters are fantastic for clean adjustments to sounds, but analogue-modelled plugins (or better yet, the real thing, if you can afford it) can add that last little bit to bring a sound out of the mix or give it a little something extra. The classic example of this is the classic Moog filter. Depending on the model and derivation of the particular plugins you have available, the two key parameters for such colouration are the resonance, that is the emphasis placed on the cutoff frequency, and the filter drive if available. The character of the sound will change with modifications to these parameters and will depend on the analogue model used. As an easy way to play around with the different modelling types, Live’s built-in auto filter device uses a few, which are also available in the filter sections of the operator, simpler, and sampler midi instruments. Ableton themselves are a little skittish about saying exactly which filters they modelled, which include the famous ms-20 and moog designs.

As a final note, I will briefly cover some of the inherent issues with filters. They cannot cut absolutely surgically, for example, a sound cannot be at 0dB at 400Hz, and cut to -∞dB at 410Hz. Filters need some space to cut, and that is called the slope. Slopes generally range from -6dB per octave to -48dB/oct. The steeper you go, the more phasing issues can occur, as the wave phase at the cut frequency range is affected by the filter. For this reason, cutting too steeply, with too many consecutive filters, or at very low frequencies can be very detrimental to sound quality. In most cases, it is not a huge issue, but something to be aware of.

Next week we will continue on this theme with how equalisers work and how to best use them.