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.

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The upper stairs has dense, bassy sound to it.

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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.

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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

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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.

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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.

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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.

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

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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.

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

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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.

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

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

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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.

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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.

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

 

 

 

 

EQ Ear Training using Max For Live

I love Max for Live, it’s DIY audio in the software domain and it makes trying out ideas and building new audio tools extremely easy.

I built this device to help improve your ability to judge and apply EQ, you will of course need Ableton Live and Max For Live to be able to use it, you can download the device here:

http://www.maxforlive.com/library/device/3657/eq-ear-training

The concept is simple; you listen to some source material and the plugin applies random EQ boosts (or cuts) and you have to guess what the frequency band is. If you do this regularly enough your ear will become attuned to the different EQ frequency bands and their effect on the source material. Just as with musical interval training, the more you do the quicker and easier it becomes to identify and apply EQ effectively, ultimately leading to better mixes / masters.

The idea came from Bob Katz book Mastering Audio: The Art and the Science , which I thoroughly recommend if you want some in-depth knowledge on some of the concepts and practices in mastering. In an early chapter he outlines various exercises to improve your ear in the context of mastering, and this is based on one of those.

I’ve put together a video to explain how to use it, it’s pretty straightforward:

 

Main features are:

  • Audition mode to hear the effect of each EQ band
  • training mode to test your ability to identify the frequency band in use
  • Apply EQ cut or boost, 3 or 6 dB and with Q factor of 0.5 (broad) to 1 (narrow)
  • Choose between your own source audio material or pink noise

Building it was relatively straightforward, there are a few quirks with max that I’m still getting used to, for instance, I only just figured out the sequence of execution of objects triggered off the same bang depends on the position of the objects in the patch editor (bottom to top, right to left).

Hope you find it useful, let me know what you think.