Preparing your audio files for working with a mix engineer / producer

Congratulations! You’ve decided that you want to take your music to the next level so you’ve hired a mixing engineer or producer. You’ve discussed how you want your track to sound, negotiated fees and worked out what particular brand of fairy dust the engineer is going to use… all you’ve got to do now is send over the audio so they can work their magic, right?


I decided to write this blog after reflecting on some of the experiences I’ve had working with audio brought to me by my clients. In some cases it’s audio they’ve recorded themselves – they may have even done a bit of editing or mixing on it too – or it’s come from another studio where they did the tracking. Either way, there is usually some work involved in getting it ready to mix.

Now I get it, as an independent artist you’re often on a tight budget and need to make sure you get the most out of your sessions. So let me help you with that right now: here are a few things you can do ahead of the session to ensure that whoever you hire doesn’t spend half the session tidying it up instead of mixing it for you.

Don’t get me wrong though, if you really don’t want to tackle any of these and are happy to pay someone to do it for you, then that’s fine. I for one will happily take on these prep tasks. But if you’d rather whoever you choose to work with can get down to doing their mix thing as quickly and easily as possible, then follow these steps:

Tracks or Session files?

Determine if the mix engineer uses the same DAW as you and can thus use your session files directly. Most engineers will support a range of DAWs:  for instance I have Pro Tools , Logic, Ableton and Reaper. If they don’t, you will have to export audio stems for each track in your project. In many ways this is easier for the mix engineer as they don’t need to worry about deciphering someone else’s project setup, but most of the following observations will still apply before you render.

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Rendering stems in Reaper

If you are exporting individual tracks, make sure each audio file is clearly named. If you have used any processing, export a dry (unprocessed) track as well as one with all the processing on. Check with the engineer on their preferred format, usually 24bit wav at the sample rate of the project. Don’t automatically normalise the stems, but watch for digital clipping and if necessary reduce the gain on the track. Logic has a handy feature which normalises the track only if clipping is detected during bounce down.

Fade / x-fade your edits

Wherever you have edited audio clips, fade-in the start and fade-out the end of the clip, or where you have edited together audio clips, ensure there is a cross fade. I can’t emphasise this enough: I’ve had audio stems delivered with clicks and pops embedded in them as the edits hadn’t been cross-faded. It’s simple to do at the time and not doing it eats up way more time and effort further down the line – time that could be spent mixing.

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Using Fades on the boundaries of audio clips in Pro Tools

Name tracks meaningfully

Your mixer won’t have worked on this project with you for the last three years, so they won’t know the channel labelled ‘Bob’s wanger’ is actually a guitar part. So name the channels in a way that makes sense, and if a channel is labelled Peruvian nose flute then make sure there are only Peruvian nose flute parts on it. Obvious maybe, but it can and does happen, for instance when you’re recording through a single input channel and adding different instruments without creating new tracks.

Clear and meaningful track naming in Logic Pro X
Clear and meaningful track naming in Logic Pro X

Group your tracks by instrument

Keep all same / similar instruments together in adjacent tracks. So, all the drum kit tracks together, or multiple layers of guitar for instance. You don’t necessarily need to group them through busses or set up mix/edit groups as this is something the mix engineer may or may not do depending on their workflow.

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Track organisation in Ableton


Assuming you know the best takes for each instrument, assemble your various takes of a particular instrument / vocal into a single comp and consolidate it, making sure you check your fades / x-fades too. You can keep the old takes on a separate track in case you need to go back in there for something later on.

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Comping with playlists in Pro Tools


Generally, get rid of any EQ / compression / reverb added on the tracks, unless there are specific sounds you want to keep or illustrate to the mix engineer (for instance that gnarly delay you love or the way that filter sweep moves). Also remember that the engineer may not have the same plug-ins as you, so if there’s something you can’t live without, freeze / commit / bounce it down to a new track so it can be incorporated in the mix. This way the engineer has both the unprocessed audio and the effected sounds to play with

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Unless it’s an integral part of the sound, remove all unnecessary plugins


Unless absolutely necessary to demonstrate a particular effect or sound, remove all automation, especially channel volume / pan / mute automation as this can really confuse things. It’s usually not immediately obvious if there’s active automation on any parameters unless you specifically view the automation lanes. Make sure the automation settings are not in write mode either.

If you do need to automate volume (e.g. to balance gain between sections), ideally use clip gain (see the next point), or place a gain / trim plug-in in the channel and automate that.

Automation in Logic Pro X
Automation in Logic Pro X

Balance clip gains

Most DAWs support clip gain – use this to balance the levels of clips on each track so that the channel faders don’t need to be moved to extremes in order to hear something, and that all parts on a track are at roughly the right volume to be heard.

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Clean it up

Remove any unused channels or muted / unused audio from the project, unless it’s stuff you think you may need later in the mixing process (e.g. that banshee wail you’re not quite sure about, or those 50 extra guitar / synth layers that you might want in the mix but can’t quite decide on yet). There’s no point in transferring 120GB of data when 110GB of that is out-takes that aren’t going to be used. This especially applies if you are transferring your files over the internet.

Use the ‘strip silence’, ‘compact’ and ‘save a copy in’ features of your DAW if they have them to remove unnecessary audio and files. Most DAWs have good file management features these days. Committing or Consolidating tracks is a good option here as it creates a single file per track containing only the audio you want to hear.

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Use Comments / Labels / Notes

Help the mix engineer navigate the song by using labels / comments / notes. This is especially important if you are not going to be attending the mix session –any accompanying notes will always be useful.

Remove any tracking-specific routing

Sometimes the audio will have been recorded in a studio with a console or other hardware for monitoring, or you may have a quirky recording set-up at home, and the project will have its I/O routing set up for that specific system. Remove the routings and set them so that all the tracks play back to a single master bus, this ensures that everything that has been recorded will be heard.

Fade / X-Fade your edits

Did I mention this? Well let me say it again, Fade / Cross-Fade your edits!

Do all this, and your mixer will be able to spend their time doing what they do best – and you can get the most out of your budget.  

Capturing the Sound of a Space

I’ve been interested in the use of convolution reverbs for a while, and was particularly inspired listening to this interview with Nikolay Georgiev by Lij Shaw for his brilliant Recording Studio Rockstars podcast series. During the interview Nikolay explains how he has fine-tuned his process of capturing the acoustics of a space using a mobile recording rig and various means of generating an impulse (including sine sweeps, a starter pistol and bursting inflated condoms!)

The basic concept is simple enough, excite the space with a burst of acoustic energy and  record the resulting response. The recorded audio is an acoustic  signature of that space which can be applied to any other sound through the process of convolution (literally ‘folding’ the sounds together).

There are two ways of performing this process, the first is the method I will explore in this blog post, using a balloon popping or other short loud burst of sound to approximate an impulse. This is the easiest method as the recorded file can be used directly in the convolution plug-in. The other method involves recording a sine sweep played back through a speaker into the space. The resultant recording needs to be de-convolved to create the impulse response. Although both methods are widely used, the sine sweep is considered better and there are some very good reasons why, but in practical terms you can still achieve great sounding results using a bursting balloon although it may not offer the most accurate representation of the space.

It was my sons 10th birthday the other day, and whilst clearing up the aftermath I found myself with a whole load of inflated balloons that needed disposing of. Perfect, I remembered I wanted to try recording some impulse responses, and this would be a great way to give it a go. I thus equipped myself with the following items:

  • Inflated party balloons (the round ones, not the sausage-type)
  • a pin
  • a Tascam DR-70D

The Tascam is a handy portable recorder that can capture up to 4 simultaneous tracks in uncompressed high quality format, with the added bonus of having 2 built-in small diaphragm condensor mics . You could equally use a computer soundcard and one or two mics.

One of the spaces I was going to capture was a small stairway that runs along the back wall of our house. I have previously had some good results using the corridor for re-amping drums , placing a mic there and playing back sampled drums into the space to add ambience.

The lower part of the stairway has a sloping ceiling which gives a bit of a flutter echo.


The upper stairs has dense, bassy sound to it.


I also wanted to try capturing the bathroom downstairs, it’s about 2m x 2m square, completely tiled floor and walls with a textured ceiling, and is quite reverberant.


The recorder was mounted on a mini-tripod and sat about 6 inches off the floor for the stairway capture, and on the window ledge for the tiled bathroom. One of the pitfalls I encountered in capturing the impulse response was that the initial pop of the balloon is very loud compared to the resulting echos and can result in clipping in the recorder. It took a few goes to get the levels just right, but once set the actual process was very simple: hit record, hold the balloon up and prick it with a pin. I didn’t experiment a great deal with the effects of location, but I generally had the balloon above the recorder when I burst it. 

Post-processing was very simple. I trimmed the impulses such that they started just before the initial transient and then faded them out shortly afterwards. I used Audacity but any DAW or editor will do the job. Once edited, just export them as wav or aiff files ready to load into the convolution plugin of your choice.

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I used the Convolution Reverb Pro device in Ableton Live Suite (it’s a Max for Live device), I like this plug-in as you can just drag and drop the IR file from finder to the device, but it’s pretty simple to do this with Space designer in Logic Pro or Waves IR-1 for examples of other convolution plug-ins.

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Once loaded into the convolution plugin, simply pass audio through it and hear how it sounds.

To demonstrate I’ll use a simple drum loop, here it is dry:

Now 100% wet with the ‘Stairway 1’ IR:

I like the nice low-end thump to that particular reverb

With 100% wet  ‘Stairway 2’ IR:

This one is a bit more present and in your face.

With 100% wet  ‘Bathroom 1’ IR:

This one has a splashy, hard character.

Notice how the character of the space is imparted on the loop. By adjusting the wet dry blend you can easily dial in as much or as little of the ambience as you like.

Another thing you can do is mess with the actual impulse response to change the character of the reverb. For example, here is the ‘Bathroom 1’ IR where I’ve altered the envelope of the sound using volume curves in Audacity:

With 100% wet  ‘Bathroom 1 Processed’ IR:

It gives it a more non-linear type of sound.

You can also apply other processing to the IR’s to create more complex and interesting effects, here is an example where the same IR has been passed through a phaser, envelope filter and finally high passed:

Although there are lots of ways you can refine the process, and there are further subtleties involved that I haven’t explored here, I was surprised how easy it was to capture the reverbs and use them pretty much straight away. I’m looking forward to getting out and about and sampling some more interesting spaces, there’s an old railway bridge near where I live that has a cool echo that I’d like to capture. So if you see a bloke with a balloon hanging out around tunnel entrances, it’s probably just an audio geek capturing impulse responses…

Tracking Vocal Groups Without Headphones

I’ve had the pleasure of working with a few vocal groups recently and have been refining the process of capturing a choir singing along to a backing track. Good monitoring is essential for vocalists to pitch and ideally in the studio we can give everyone headphones, but not many places can cater for a large vocal group unless your budget stretches far enough for a place like this.

For the Vocal Works Gospel Choirs last album (Fourteen), we tracked in the big room at Real World Studios and hired in 50 wireless headphones from Silent Disco, which worked a treat as it gave everyone independent level adjustment, avoiding spill and  the cable chaos a wired setup would have caused.

For those moments when you don’t have enough headphones, and it’s not worth hiring in a load of gear, there are still ways to get great results.

In this example for Bath Academy of Musical Theatre, the singers had an instrumental backing that they wanted to record over. We didn’t have the facility to give everyone headphones so the setup I went for was quite simple in the end. I used a pair of AKG C414 mics in Blumlein configuration, placed above a monitor speaker with the choir arranged in a circle around the mics.

Now the room you record in will have a big influence on the sound, we recorded this in a relatively dry and well proportioned room (a modern music department in a local school) which really helps by reducing the amplitude of reflections from the walls. This is particularly important when you are playing the backing track into the room whilst recording as it reduces what I call indirect spill which will add unwanted ambience to the backing track.

Placing the monitor speaker below the mics puts it in the most off-axis position in the Blumlein pickup pattern, so you get attenuated direct sound from the monitor speaker into the mics (though this does not eliminate the playback sound from entering the mics), and the monitor ideally should not have an upwards facing driver. Also, you need to set the monitor volume to the lowest you can get away with for the vocalists to feel happy and pitch well.

In the video example, we recorded the soloists individually afterwards using headphone monitoring, but all the ensemble parts (from 2:22 onwards) were recorded live along with the backing track in the manner described. When it came to mixing, I was impressed at how much easier it was to get the backing track to sound good with the choir. In previous approaches I was never completely happy with how the vocal mics added a room sound to the backing track, especially when you start compressing them.

As a way of quickly and easily tracking a vocal ensemble with minimum hassle, this definitely works. I have also come across similar examples of this used in studio recordings, such as using a figure of 8 pattern mic with singers either side on-axis, and a monitor at 90 degrees in the maximum off-axis position.

Modifying the Rode K2 mic Part 1: Tube Change

I originally intended to replace both tube and capsule on this mic, but after experimenting and comparing I ended retaining the original capsule but with a replacement tube. In this post I’ll describe the simple tube change and give some audio examples. In the next post I’ll go over the capsule change. To be clear, the capsule upgrade made the mic sound different, rather than better, and based on what I already had in my mic locker I chose to return to the original capsule.This highlighted the importance of following your ears rather than what you read on the internet!

Okay, so here’s what you need:

A Rode K2 valve microphone ( I got mine second hand off EBay for £300 which is a fair price). These are decent mics and are very useable straight out the box. They are multi-pattern, very well made with a good quality power supply, cable and shockmount (unlike the Chinese valve mics often used for mods such as the Apex 460). The case is a bit tacky but you can’t have it all.

A NOS (New Old Stock) Telefunken E88CC (6922) valve or equivalent


I learned a lot about the vintage tube market whilst researching these. Firstly, they are not cheap! You have to be careful as there are a lot of rip-offs out there, people trying to pass off newer tubes as vintage ones but fortunately there is a wealth of info out there to help as well as several reputable dealers that get recommended by many different people. I paid approx £125 for one from NOS Tube Store , their service was great and the tube was as described. This particular one is a sought-after Telefunken made in West Germany in the 1950’s. There are many other fine vintage tube options but I didn’t have the time and money to experiment, and this tube was highly recommended by a lot of people. If you want to know more and have time to burn, just check out the forums. Different tubes will impart a different sonic character to the mic so the ultimate test is to use your ears!

How to do it

This is very easy and I would recommend you try this first and see if you like the way it sounds. I recommend you record some different sources with the stock tube before you proceed, so you can compare afterwards and understand for yourself how the mics character has changed.



Firstly, undo the metal retainer ring at the base of the mic and remove it. Then undo the metal sleeve that forms the main body of the mic (hold the mic by the grill end and unscrew). It should then slide off to reveal the tube and circuitry of the mic as shown in the picture.


The tube is secured in position by a plastic clip which sits on top of the tube (pointy end) and the clip is held in place by 2 screws. Using a Phillips screwdriver, loosen both these screws and fully undo 1 of them to flip the plastic clip up and out of the way.


The tube can now be pulled carefully from its socket, a slight amount of wiggling (of the tube, not you) may help!


Carefully insert your new tube into the socket, it only fits in one orientation, make sure it is fully engaged. Replace the plastic retaining clip over the top of the tube and tighten the screws.

To my ears the NOS Telefunken tube makes the mic more mid-focused, the low frequencies are still clear, warm and big, the highs are a little less bright and sharp but the mid presence makes the mic cut through the mix more. I also find it more pleasant to listen to than with the stock tube, but it may not suit every application. I’ve presented some comparisons here to give you an idea of the difference. All clips were recorded via the preamps on my UA Apollo interface, I’ve endeavoured to maintain the same conditions and distances between mic and source for each comparison, but there may be small variations.

Firstly, male voice (don’t worry, I’m not singing)

Stock Tube

NOS Telefunken E88CC

Female voice

Stock Tube

NOS Telefunken E88CC

Double Bass

Stock Tube

NOS Telefunken E88CC

Judge for yourself and decide if this mod is worth the money, for me it’s a yes, but I will qualify it by saying I have other flavours of mic tonality in my locker, this adds something different, and it does it rather well.

In the next post I’ll describe changing the capsule and post some examples of the difference that makes.





AKG C414B Hum problems

I thought I’d write a quick post on this in case some of you experience a similar problem, it may save you hassle and expense.

I have a pair of AKG C414B mics, they are real workhorses and as the cliche goes, the studio Swiss army knife. On top of that they are solidly built and very reliable. But sometimes things go wrong. I recently did a bit of drum recording for myself and was listening back when I noticed a faint but distinct mains frequency hum on the overheads (my 414s). I had them running through my home made Neve preamps and then into a pair of line-level inputs on my UA Apollo interface. I automatically assumed the hum was coming from my home-made preamps rather than the solid, Austrian-engineered AKG mics. I duly set out to test, swapping cables, microphones and trying various configs of phantom power on / off on each channel, until it was clear that the problem lie with one of the 414s.

Faced with an expensive repair, I thought, well, let’s open it up and see if anything obvious was up with it.

Hmmm, lots of surface mount components and no obvious signs of trouble. A quick Google search led me to this post. Not exactly the same problem but similar, so I read on. Turns out that the grill / mesh of the mic connected to ground via pin 1 of the XLR forms a Faraday Cage, which helps shield the high-impedance capsule from EMI noise in the room. The mesh makes contact mechanically and if this connection is a little dodgy, it won’t work and there will be noise. I tested the resistance between pin 1 on the XLR and the mesh, it was a variable which immediately suggests a problem. I went about gently squeezing the base of the mesh in an attempt to improve the mechanical connection. Checking again, the resistance was now consistently minimal, so time to test. I put the body back on and plugged it in, powered it up and hey presto, no noise. Phew!…. So if you’re having this kind of issue, try this first, the mic is easy enough to open , just remove the 2 screws on the base (star driver or a flat head screw driver will do) and the smaller cross head screw in the XLR connector base and then slide the body off.

Flying mics and ominous cymbals

I recently read an article on the flying mic technique – not sure where, may have been TapeOp – but being a fan of the more esoteric side of recording, I thought I have to try that out.

The idea is simple, you suspend a mic from a pulley via a length of string which allows you to smoothly raise and lower the microphone. You then record a source and vary the mic position as the source sound decays, allowing you to pick up more of the detail in the decay of the sound.

This technique is particularly interesting with cymbals. If you have the mic too close, the initial attack will likely overload it so you won’t capture the detail in the moments immediately after the transient, but with the mic far away you can’t readily pick up the fine details of the decaying sound. And it’s in that decaying sound that a lot of the interest lies; cymbals produce lots of non-harmonic tones, i.e. frequencies not related to each other nor a fundamental pitch by integer multiples as we find with strings or pipes.

I set up a mic stand over my ride cymbal and improvised a makeshift axle from the clamp for a reflection filter, this would function as a pulley and allow a length of string to travel freely over it and up and down over the cymbal. For the microphone I chose my trusty Oktava MK-012  (modified by Micheal Joly) with an omni-pattern capsule so that I could get really close into the cymbal without the proximity effect. I tied the mic to one end of the string and made sure there was enough free XLR cable to allow the mic to travel freely up and down.  With the other end of the string in my left hand and a drum stick in my right, I practiced striking the cymbal whilst simultaneously lowering the mic. It took a few goes to get the timing right but eventually I could ride the mic in just after the initial hit and get it very close to the cymbal surface for the decay.

I also experimented with moving the mic around the cymbal surface to capture different harmonics. Monitoring on headphones also allowed me to get the movement right in terms of riding the volume.

So here are some results:

I think there’s some great scope for sound effects and soundscapes using these. With that in mind, here are a pair of hits reversed, hard panned and with an added splash of reverb to create an eerie, ‘something bad is about to happen’ kind of sound:

Happy experimenting…..


“Boutique” DI box on a budget

I’ve been meaning to build a passive DI box for sometime, I have a couple of active ones and it’s nice to have an alternative. Also, I recently bought a Korg Minilogue and figured a good quality passive DI would be a good match.

Passive DI’s are very simple, they are essentially a step down transformer used to bridge the high impedance output of an instrument to the lower impedance input of a mic preamp, as well as to convert from an unbalanced to balanced signal. As they are based on a transformer, the performance of the DI box is pretty much uniquely governed by the transformer, and good quality transformers aren’t cheap. There are a number of high quality DI boxes out there, such as the Radial JDI, but being a DIY fiend and a slight cheapskate, I thought I could have a go at making one, how difficult (and expensive) could it be?

There are a few great posts giving instructions on building a DI box from scratch, but what I noticed was that as you get all the parts together, the bill of materials creeps up and the cost benefit of doing it yourself  becomes less attractive. Now, cost is not the only reason why I build things, in fact, those EZ1290 preamps ended up costing me way more than I expected (they were a labour of love), but frankly, if I can build 2 good quality DI boxes for the price of buying one, then that’s worth the effort for me.

Whilst doing my research I noticed there were a number of cheap passive DI boxes out there, and I’m guessing the reason they were so cheap was because they used a poor quality transformer, so rather than build a box from scratch, why not just upgrade a cheap one? I found this Cobra DI box on EBay, it only cost £9.95 (+£2.95 p&p) but comes in a metal case with pad and ground lift circuitry and a thru jack. Perfect as a donor, all I needed was a transformer.


I went for this Cinemag CM DBX (EBay £64.99)


Good alternatives include:

Jensen JT-DB-E/EPC (as used in the Radial JDI)

Lundahl LL-1530, LL-1576/77/78

Sowter 4243, 8044

The Cinemag website includes schematics for a DI circuit using their transformer, comparing this to the COBRA DI box there were a few components missing but I thought I’d wire it up as a direct replacement for the existing transformer first and then see if there were any issues.

Firstly, open up the COBRA DI box by unscrewing the 4 retaining screws on the base.


With the cover removed you can see the simple and robust construction, the original transformer is through-hole mounted on the PCB.


Flipping the box over and working with a solder sucker, remove the original transformer.


You can see the obvious differences between the original and upgrade transformer, the mu-metal can for shielding, but the Cinemag also has individual shielding for each winding and will use better quality wires and laminations.


I soldered the Cinemag wires into the existing PCB holes following this wiring scheme:

  • Yellow – Input Tip
  • Orange, Black, White – Input Sleeve (PCB ground)
  • Gray – XLR 1
  • Red – XLR 2
  • Brown – XLR 3

I tested this configuration and it worked fine, but I noticed that in the Cinemag schematics they wire only the white cable to case ground (not the black and orange too), so I modified the wiring slightly such that black went to the PCB ground and white connected to the case via the existing green cable.

I will admit that I was feeling lazy and didn’t fancy drilling the case or the PCB to mount the transformer, so instead I used cut up neoprene pipe insulation to snuggly hold the transformer in place between the case and PCB. Once the case is on the transformer isn’t going to move anywhere and there is no strain on the connectors.


And that’s it, job done, a very simple and quick route to getting a high quality passive DI box on a budget.