Room Resonances

Being able to identify room resonances, and then work with and around them are key to producing balanced mixes.

Most of us working in project studios, are mixing and producing in environments which are far from acoustically perfect, and having to deal with frequency peaks and nulls in different parts of a room are an unfortunate but unavoidable reality. Being able to identify room resonances, and then work with and around them are key to producing balanced mixes.

I faced room resonance issues when mixing my most recent track.  My studio room is far from acoustically ideal, with concrete walls (although covered on 3 sides) and almost-square dimensions (apart from a corridor at the back, forming an overall ‘L’ shape).  My normal sitting position when mixing is centred in the room, and forms an equilateral triangle with the monitors (as is recommended by many tutorials, and monitor instruction manuals).  In the past this position has always sounded balanced in terms of frequency response, but with the last track, i was finding that the mix sounded more balanced when i sat about 50cm in front of my normal position… but as soon as i moved back, the low end of the bass line dropped out significantly.  The bass line centred around a D note (approx 73Hz), and after messing around with sine wave sweep tones, i found that there were significant nulls at that frequency in my normal listening position, and other places in the room.

As a test, I played at 73Hz sine wave through the monitors, and recorded clips of it at two places… one where i thought the mix had previously sounded reasonably balanced, and another where the sine wave seemed to drop off the most (both points being equidistant from the speakers).  These two clips are below (note… please make sure you’re listening on something that can play back 73Hz, or you’re not going to hear anything!)…

Null point:

Balanced point:

Despite the fact that the recordings are of exactly the same sound recorded at the same distance from the speakers, the clip recorded at the null point is roughly 6dB quieter than the clip from the other point.  I was surprised by this… 6dB is really significant, and I assume that the difference between the null point and a peak point in the room could be even as much as 12dB.  If you inadvertently did your whole mix from the null point, it would potentially end up 6dB too loud around 73Hz… that’s a big difference, and would sound noticeably unbalanced when played back on other systems.   It would have been especially problematic in my case given that the null frequency, and the fundamental of the key of the track were the same.

Identifying null and peak points is the first step , and the next question is how to work with/around it?  In my case i changed my listening position slightly, shifting about 40cm forward of the normal position.  I knew from mixing other tracks that this spot usually sounded slightly bass heavy and a little dull at the top end (as it was slightly off-angle of the monitor tweeters).   So I had to be conscious of this when mixing, and very  slightly compensate for it… mixing to be slightly more light in the bass and crisper at the top end than what i thought was an ideal balance.  I also occasionally moved back to the normal position in line with the tweeters, but just to evaluate just the high frequency content.  I also regularly checked the mix on other systems to get some additional perspective (my old monitors plus my tablet and earbuds).

In the end I achieved what I think is a nice, balanced mix through adjusting the mix position as described, and manually compensating for the deficiencies in frequency response at various positions.  This was also coupled with other techniques (which I’ll describe in detail in a future post).  It also helps enormously to ‘know’ the sound of the room you work in… to know and remember any null and peak points, and to be able to anticipate the effect they will have on different parts of a mix, and compensate and balance accordingly.  When I was only producing music in my free time, I didn’t notice the effect of room resonances as much… I think producing full time, and working in the same space regularly lets you get to know the sound of a room much more quickly, and be more conscious of any differences or anomalies.

Interestingly, I checked the wavelength of the low D in which the key was based, and found it was just over 4 metres… which was almost exactly the length of the back wall of the room… and hence probably explained the peaks and nulls at that frequency.

Cleaning Up a Mix

there’s usually not one magic fix in order to realise a fairly abstract goal like ‘make the mix clearer’

Over the last week I’ve been finalizing the mix of a new track (Summer Wave).  In terms of sound texture, it’s the ‘thickest’ track I’ve written this year, with quite a lot of instrument and percussion layers mixed together.  The thicker the texture of a track gets, the more challenging the mixing process becomes, as you’ve got more layers of sound, and more frequencies competing to be heard in a limited space.  Hence, early in the process when I started with a rough sequenced mix, one of the first things I wanted to do was clean up the mix… to remove ‘mud’ and make the individual layers more distinct and audible.  Generally I find that in removing ‘mud’ from a mix, there’s usually no one ‘silver bullet’ solution, and the improvement comes from repeated iterations of small fixes.  That was the case here, but there were 2 changes which both made a significant improvement to cleaning up the mix.

The rough mix sounded like this…

…not too bad for a first cut, but i wanted the individual elements to be clearer.  While doing some cleanup work on some of the individual layers, I soloed this ‘glass bottle’ track (so named because it came from a sample of a glass bottle being tapped on a tiled floor)…

I was surprised at how much low frequency content there was in this part… especially because i usually high pass filter the raw samples of sounds like this long before I get to the mixing stage.  The sample had a loud transient ‘thud’ sound at the start at approx 135Hz.  This sat right in the frequency range of both the bass line and the ‘meat’ of the bass drum, and given the ‘glass bottle’ sound had been included for its high frequency, bell-like rhythmic pattern, this sound down around 135 Hz was redundant, and was probably just ‘muddying’ the sound of the bass drum and bass line.  I initially applied a high pass filter at ~300Hz, but after a few more iterations of review decided I could set it at 518Hz without detracting in any way from the part of the glass bottle sound I wanted to hear.  The soloed glass bottle sounded like this with 518Hz high pass filter applied…

The full mix after this change, sounded like this….

Granted its subtle, but to me there’s a definite improvement in the ‘smoothness’ of the bass line (because the rhythmic pulsing at around 135 Hz caused by the glass bottle pattern has been removed).  And importantly, as discussed at the start of the post, it’s an important step in the iterative process of cleaning up the overall sound.  (Note – to more clearly hear the ‘smoothing’ in the final full mix, download the before and after mix clips and A/B them with a low pass filter at about 200Hz).

More towards the end of the mix process, i was reasonably happy with the overall sound of the mix on my monitors, but i felt that the synth ‘stab’ sound was not clear enough in the mix when auditioned through my tablet and earbuds.  The mix at this point sounded like this…

After soloing some of the parts, i realised that one of the background percussion parts (sourced from a sample of an aluminium coke can) had a note which played at the same time as the synth stab…

Coke can…

Synth stab…

The problem was that the fundamental of that first coke can note was at 221Hz (the A below middle C), and that same A was one of the notes in the synth stab chord.  Basically the 2 sounds were competing for the same frequency space.  Give that first note of the coke can was really just a grace note to the second higher and more prominent note, I made a 3.3dB cut at 221Hz on the coke can track, which resulted in…

And sounded like this in the context of the whole mix…

To me this made a pretty significant contribution to allowing the stab sound to sit more clearly in the mix.

Again, my experience is that there’s usually not one magic fix in order to realise a fairly abstract goal like ‘make the mix clearer’.  But through iterative and successive iterations of fixes like those above, high-level overall improvements can be achieved.



Compression Basics – Compressing live percussion

Live percussion recordings tend to have a large dynamic range, and hence are a great vehicle to use to learn the basics of compression.

Compression seems an appropriate topic for my first ‘howto’ article, given that it’s the effect that I’ve learned by far the most about over the last 6 months.

I tend to use a lot of recordings of live sounds in my tracks, particularly for percussion.  Live percussion recordings tend to have a large dynamic range, and hence are a great vehicle to use to learn the basics of compression.  Because the dynamic range is so large, it requires either a single pass of a compressor with very aggressive settings, or better, successive applications of more gentle compression and limiting (as I’ll show here).

Applying compression is often a difficult technique to learn, because the differences imparted by mild compression (e.g. with low ratio or high threshold, for example on a master bus) can be difficult to recognize unless you really know what to listen for.  However, when reducing the dynamic range more dramatically (as is usually required on live percussion samples) it’s much easier to hear the effects of the compression

The example sound I’ll use is a recording of a steel drink can knocked against a hard table (recorded using a Rode NT3 condenser microphone). I thought it was an interesting sound and wanted to save it in my sample library so I could potentially use it in a track at some point in the future.

One important point here is that you’ll need to use good headphones or monitor speakers to properly hear the difference between the audio samples below.  It will likely be difficult to hear the differences properly on laptop, tablet, or phone speakers.

The raw sample sounds like this…

and has the following waveform…


…straight away you can hear (and see) the big difference in level between the initial transient sound of hitting the table, and the ‘tail’ sound of the can ringing (starting from about 0.015 seconds).  If you tried to use this sound in a track as-is, you’d have to keep the level of it fairly low to prevent the loud transient from clipping, and then the nice, harmonic ring of the can would probably be completely lost under other sound layers.

For these types of samples, I usually firstly apply some limiting to reduce the level of the transient peaks (using Waves L1).  In these cases I often find looking at the waveform more closely helps to give you an idea of where to initially set the threshold of the limiter…


In this case, I ideally want to trim the two most prominent peaks at the level marked by the red lines.  These show a 16bit integer value of 18,000, which equates to roughly -5.2dB (note that waveform axis is marked as 16 bit, although the sample itself is 24 bit).  Auditioning L1 on the sample, I was actually able to limit down to -8.2dB threshold without adversely affecting the sound.  Also, because we’re just limiting peaks in this case which rise and fall very quickly, I’m using a very short release value.  Ultimately I used the following settings in L1…


… and it resulted in the following sound and waveform changes…


Zooming in on the waveform again, what I want to achieve is to further reduce the difference between the peaks and the ring of the sound… visually, to try and ‘pull’ the peaks more towards the red lines.  Using limiting again would be too harsh, and would probably remove the dynamic and ‘impact’ out of the sound… hence I use a compressor (Waves C1).  Again, using the red lines as a guide for the initial threshold setting, these are at 16 bit value 6000, which equates to approx -14.7dB.  I probably want to reduce the level of these peaks by about a 1/2 or a bit more above the threshold, so I would guess at a compression ratio of around 2:1 to 2.5:1


As with the limiter, I’m trying to just ‘pull down’ transient peaks here whose wavelength is very short (a handful of 44.1Khz samples), so I use short attack and release settings in C1.  From previous testing, I’ve found the quickest attack and release settings you can use in C1 without it introducing undesirable artifacts (‘clicking’ sounds as the compressor engages) are about 0.04 and 30ms respectively.

Ultimately I used a bit higher threshold than the -14.7dB estimated.  The reason for this that C1 has a fairly soft ‘knee’ (i.e. it starts introducing compression gently as the sounds approaches the threshold level).  I looked at the waveform to get a rough idea of the initial threshold and ratio settings to use, but these need to be auditioned and finalized by ear.  I settled on the below settings, which gave a nice balance of still having some dynamic and ‘impact’ but allowing the ‘ring’ part of the sound to be closer in level to the transient (it showed around 3dB of gain reduction on the meter in C1).  The final step was to add 1.9dB of makeup gain, which audibly level-matched the compressed sound with the original.


If I was using this sample immediately in a track I probably would have gone for slightly more aggressive settings (less threshold or more ratio), but given it’s to be put in my sample library, I erred towards more conservative settings to make the sound more generally useable.  The resulting sound and waveform are below…


At this point I normalized the level of the sample up to -3dB.  The final step I usually take with these kind of samples is to do one more application of L1, just to trim the highest peaks, but without changing the sound of the sample.  This is just to try and reduce the transients as much as possible, which makes the sound easier to mix into a track without master bus clipping (I’ll discuss in more detail in a later post).  Looking at the waveform again to give me a guide for the initial settings, I want to try and contain the peaks to within the red lines (16 bit value of 20,000 ~= -4dB).  I used a threshold of -4dB in L1…



…which trimmed the peaks, but without noticeably changing the sound.

With these types of percussive samples, the last step I take is usually to use a gate to fade out the tail of the sample.  The appropriate gate settings are best judged by ear, and I settled on those below, which I thought gave a nice balance between allowing some of the ‘ring’ of the can to sustain, but also fading out the hiss of the noise floor (and I used the Waves C1 Gate for this)…


If you compare the initial raw sample against the final one below, the final one has a lot more evenness between the transient and the ‘ringing’ tail of the sound… the whole of the sound can be heard more clearly, and this will make it far easier to mix into a track along with other instruments.  It has a ‘stronger’ sound than the raw sample, but peaks at a lower level.

For readers who are a bit unsure of the appropriate applications of compression and what settings to use (as I once was), I’d encourage you to try the above steps with your own live percussion samples.  For me it was a really good way to practically understand the effects of compression, and to be able to clearly hear the results.

First post, and Welcome…

Hello… you’ve arrived at the first post of chromaticsabatic!

Hello… you’ve arrived at the first post of chromaticsabatic!  The goal of the site is to provide practical ‘howto’-type advice on all aspects of writing and producing electronic music, backed up with practical, real-world examples from my own tracks.  I hope to cover technical aspects like sequencing, mixing, and applying effects, as well as the more artistic aspects of the writing and production process.

I’ve been writing electronic music as a hobby since high school, and although I’ve always gotten a huge amount of enjoyment and satisfaction from producing my own stuff, trying to fit music in with full time work never allowed me to focus enough to achieve the quality of production I wanted.  So, at the end of 2015, I left my day job to write music full time. Right now I’ve had just over 6 months of being a full-time musician and producer… it’s been a challenging experience, requiring dedication and perseverance, but at the same time hugely rewarding to be able to immerse myself in something I love doing.  Through hard work and a lot of trial and error, I’ve learned a huge amount about the music production process… particularly with regard to mixing (EQ, compression, spatial effects), and final ‘polishing’ of a track to get it sounding as close as possible to commercial releases.  As the year’s progressed, I’ve documented a lot of what I’ve learnt for personal reference, but being able to share this knowledge with a wider audience will make the whole experience more worthwhile.  Hence I’ve started this site to give me a vehicle to share the techniques I’ve learned, and to continue to document things I discover during the rest of the time.

As far as music styles go, I try not to confine myself to specific sub-genres of electronic music, but what I’m writing at the moment sits somewhere between house, progressive, minimal, and techno styles.  Samples of my music are available on my soundcloud profile.

My greatest satisfaction from this process, will be if readers can use the techniques to develop and improve their own music.  I’m really pleased to be able to host this site, and I hope you can get a lot from it.

Alastair Wyse