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Creative Convolution: New Sounds From ImpulseResponsesTips & Techniques

Technique : Processing

When is a reverb not a reverb? When it’s a filter, ofcourse! There’s more to convolution than meets the ear,and creative processing of impulse responses can yieldextraordinary results.

Emmanuel Deruty

he common perception of convolution reverb plug-ins, which are based on the use of impulse responses, is that theyoffer great realism, but limited potential for experimentation. In this article, however, we’ll be exploring the creative sideof convolution. Being audio files, impulse responses can be edited, modified and even created from scratch, and we’ll

see how this opens up new ways of processing sound. I’ll be proposing several innovative methods for impulse responseauthoring and processing, and offering an original understanding of reverberation.

This article is accompanied by a number of audio examples, which are available on-line at www.soundonsound.com/sos/sep10/articles/convolutionaudio.htm. The audio examples are all numbered, so I’ll refer to them simply by their number in thetext. Most reverberation examples are based on a single dry audio file that was recorded inside an anechoic chamber (seeaudio example 1). I have also provided a number of impulse response files corresponding to the relevant audio examples, as24-bit, 44.1kHz WAV files.

The Spirit(s) Of Reverberation

Impulse response convolution is best known as a technique for adding reverberation to a given sound in a realistic way. Thatsaid, realism in terms of reverberation is not always an easy concept to define, reverberation being a complex and multi-faceted phenomenon. St. Peter’s Basilica produces reverberation, and so does your shoe cupboard: two transformations thathave little to do with each other from a perceptual point of view, though they both qualify as reverberation.

Moreover, when we’re using reverberation in a mix, realism isn’t necessarily required, nor is it always welcome. Certainly,reverberation can be used to give recorded music a sense of being performed in a real acoustic location; but it can also beused to provide an acoustic environment to a given track so it stands apart from the mix. In this case, realism in thereverberation can be welcome, but it’s not the main purpose. Going further, one can add reverberation purely for sound-designpurposes, in which case realism is not an issue.

As we can see, there are many points of view from which to consider reverberation. In this article, we’ll forget about realism,and remain open to the idea that an impulse response (or IR for short) recorded from a cardboard tube and then processeddigitally using EQs or any other audio tool provides as interesting a timbre as one recorded realistically from Vienna’sMusikverein venue. We will study impulse responses and convolution plug-ins for their ability to transform sound and createparticular timbres. Convolution reverbs provide a convenient way to extend the notion of reverberation to new limits, anddiscover sounds that were previously unheard — and that’s what we’re after here.

What’s Possible With Convolution

To master the process of impulse response convolution, it’s important to be aware both of what it can do and what it can’t do.Impulse responses are audio files that contain information about audio transformations. They’re designed to be used inconvolution engines that know how to interpret this information. These can be of two kinds: reverberators and filters. It’s veryimportant to understand that impulse responses are not able to capture anything else. They’re just not the right tool to capturecompression, distortion, pitch-shifting or modulation effects. For instance, trying to generate an impulse response from aManley Variable Mu compressor just doesn’t make any sense. This unit’s interest lies in the compression’s attack and releaseshapes, as well as in the fantastic colour of the harmonic distortion it generates when pushed hard. Neither of these featurescan be captured inside an impulse response file.

To get a better understanding of the issue, it may be useful to consider to what extent an impulse response is able tocapture the sonic characteristics of a hardware EQ — not a theoretical filter, but an actual analogue outboard EQ, which doesmuch more than frequency filtering. Making an IR from such a unit can make sense, but it will leave out a number of the unit’sintrinsic properties, such as harmonic distortion, background noise and dynamic behaviour. This would be particularly true of abudget tube EQ such as the Aphex 109. This sometimes generates quite a lot of what seems to be intermodulation distortion,softening the sound considerably and making the unit unique, in a nice way. When taking the impulse response of a 109, youcan forget about the uniqueness: what you will get is a plain, digital filter. Not really an EQ, just a filter. It’s not the same.

Never forget that impulse response-based processing is, in essence, digital. What’s more, it’s strictly deterministic. If youplay a given sound inside a convolution plug-in, you will always get exactly the same result. This would not be true in the caseof acoustic situations such as reverberation from real rooms, in which you never get the exact same response twice, no matterwhat.

Reverberation Basics

Before actually dealing with convolution reverbs, let’s take a closer look at reverberation itself. Acoustic reverberation can bedivided into two stages: first or early reflections, and the diffuse field that comes after these. Early reflections are easy tounderstand: imagine that you’re at your desk, in front of your laptop. Reading this article, you may very well be in this precisesituation. If you speak, the sound of your voice will strike your computer screen and keyboard, and bounce back to youdirectly. These reflections are discrete, as opposed to continuous: they can be identified individually. In an impulse responsefile, they appear as peaks, or clicks.

The diffuse field is less easy to understand. Imagine you’re shouting in a big factory. The sound of your voice will strike thesurfaces around you, creating the first reflections. In turn, these reflections will be reflected and diffracted on other surfaces,creating other reflections, which will also be reflected and so on. Eventually there will be so many reflections that the resultingsound will become continuous. This continuous stream of reflections makes the diffuse field. In an impulse response file, thediffuse field appears as a continuous noise.

In this article:The Spirit(s) Of

ReverberationWhat’s Possible With

ConvolutionReverberation BasicsDiscrete Or Diffuse?The Difference

Between ReverberationAnd Filtering

The ContinuumBetween Reverb AndFiltering

Other Ways To ModifyIR Length

IR Length: Altiverb VsPeak

Diffuse FieldAuthoring With Noise

Percussive SamplesAs IRs

Speakers &Headphones As IRs

Space DesignerPrograms

ConclusionCreative Impulse

Response Recording

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This preset from the TC Electronic M3000shows a clear distinction between earlyreflections and diffuse field.

A different preset from the same unit, bycontrast, shows a gradual transitionbetween early reflections and diffusefield.

Diffuse field in the impulse response froma bedroom closet.

In actual rooms, discrete early reflections gradually become too numerous to be discriminated individually, thus turning intoa continuous diffuse field. There is no clear limit between one and the other. However, the distinction is useful to make, inorder to understand how reverberation works. Also, some non-convolution-based digital reverb algorithms use separateprocesses to generate early reflections and the diffuse field, with no gradual transition from one to the other. If you take animpulse response from such an algorithmic reverb, the two are clearly distinct. For instance, the impulse response whosewaveform is shown right was recorded from the TC Electronic M3000 using preset354, ‘Mine Corridor’. There is no smooth transition between early reflections anddiffuse field. Both can be clearly distinguished.

The second impulse response was recorded from the same unit, this time usingpreset 305, ‘Wide Garage’ . By contrast with the first example, early reflections aregradually turned into diffuse field, and it’s difficult to distinguish a clear point atwhich this happens.

Intuitively, one would think that only large spaces, such as cathedrals, produce adiffuse field. Indeed, if you clap your hands in your bedroom, it’s unlikely that youwill actually hear a clear diffuse field, such as the one you can hear in a warehouse.But in truth, in any acoustic space, there is always a diffuse field. It’s just that insmall spaces the reverberation is too short, everything happens too quickly, and it’simpossible for the ear and the brain to discriminate the early reflections from thediffuse field. For instance, consider the impulse response whose waveform isshown on the right , recorded from a bedroom closet. This is indeed diffuse field! Noearly reflections are apparent. Refer to audio example 2 to hear the result of suchan IR, and to obtain the corresponding IR file.

Discrete Or Diffuse?

To get a better understanding of the transition between discrete reflections anddiffuse field, it can be useful to carry out a simple experiment. For this purpose, let’sconsider a derivative of what’s called a Dirac Impulse: a short impulse, which canbe digitally rendered as a single ‘1’ in a series of ‘0’s. The screen overleaf showssuch a waveform in BIAS Peak.

Considered as an impulse response, this Dirac Impulse corresponds to a singledelay, which does not colour the sound at all. For those versed in mathematics, thisDirac Impulse is the ‘identity element’ of convolution, and as a consequence, ofimpulse responses. Now, if we add four other Dirac Impulses to this first one, we willget an IR with five reflections, corresponding to five discrete echoes. The moreDirac Impulses we have in an impulse response, the more reflections we get. Let’scontinue adding Dirac Impulses to each other in a random way, until eventually weend up with an impulse response that contains 32,000 reflections. An interestingphenomenon will then happen: above a certain number of reflections, the ear willnot be able to discriminate between the reflections, and will begin to hear a diffusefield.

Refer to audio examples 3-15 to hear the transition from discrete reflections todiffuse field. Example 3 is based on five Dirac Impulses within a one-secondtimeframe, example 4 on 10 Dirac Impulses, and so on up to example 15, which isbased on 32,000 Dirac Impulses. The corresponding IR files are also provided. Ifyou listen to the IR files themselves, you’ll hear that discrete Dirac Impulses seemto completely disappear when there are more than 4000 of them. When listening tothe audio examples, a diffuse field begins to be heard above 500 simultaneousDirac Impulses.

The Difference Between Reverberation AndFiltering

Let’s consider the perceptual disappearance of the discrete reflections and theadvent of the diffuse field in the example above. This phenomenon happens at between 200 and 500 discrete Dirac Impulsesin a one-second span. This would suggest that the human ear is able to discriminate phenomena that are separated by aperiod of 50ms, but not by a period of 20ms. Consequently, it means that there is a time constant above which the ear iscapable of discriminating consecutive events, and below which it is not.

This constant does indeed exist: it’s called the ear’s integration time. It’s a very important notion as far as hearing isconcerned. To make this notion perfectly clear, consider two extremely short audio samples, such as digital clicks. If those twoclicks are played back with a one-second interval between them, they can easily be discriminated. Play those clicks with a100ms interval between them and it’s still possible to hear two clicks, but it’s less easy. Play those two clicks a mere 10msapart and it’s impossible to hear two clicks: they are perceptually merged with each other. In place of the two clicks, we hearone compound sound. The time interval over which those two clicks can be discriminated is called the ear’s integration time,and is described in the technical literature as around 40 to 50 milliseconds.

This is a very important phenomenon. It is what makes us able to hear pitch instead of consecutive sound events: 50mscorresponds to a frequency of 20Hz, which is the lower range of human audition. In the context of reverberation, it is whatturns discrete reflections into diffuse field. It also brings another interesting consequence. Consider an impulse responsearound 100ms long: perceptually, this impulse response corresponds to a reverb. Then, consider an impulse response that isonly 10ms long. This impulse response is not perceived as a reverb — as something that possesses an existence over time —but as a filter, something that is perceived as being instantaneous.

Quite importantly, this means that as far as impulse responses are concerned, reverbs and filters are the same thing, basedon the same content. Only perception makes them different. In this article, it means that by dealing with convolutions ingeneral, we’ll deal with both reverberation and filtering.

The Continuum Between Reverb And Filtering

Now that we know that the only difference between reverb and filtering is the span of time over which the phenomenonhappens, we’re going to convert a reverb into a filter, by gradually reducing the length of its impulse response.

For that, I’ve used the Change Duration function in BIAS Peak, but we could have used any other time-stretching algorithm.We start with the impulse response recorded from a small indoor pool. It lasts about two seconds, and features a clear-sounding reverb, with harmonic normal modes developing near the reverb tail. We process this impulse response repeatedly,each time reducing its duration by 50 percent. After five or six passes, the impulse response loses its reverb aspect. At thispoint, its duration is respectively circa 60ms or 30ms. After 11 passes, its duration is 1ms, and it’s definitely a filter, with afrequency response that seems to correspond with the averaging of the initial reverb’s frequency response over time.

The audio examples corresponding to this experiment, along with the corresponding IRs, are numbered 16 to 27. Listen

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A Dirac Impulse in BIAS Peak.

Altiverb 6’s Reverb Size and Time

through them in sequence and you’ll hear how the room sound seems to shrink, eventually turning into a filter that keeps somecharacteristics of the initial room.

This is an interesting experiment, but it also provides an interesting range of unusual impulse responses that can be of greatuse in production: a variety of small spaces, along with interesting filters that would be hard to get on an actual EQ.

Let’s try that again, this time starting with a completely synthetic IR, processed from a pseudo-periodic oscillator. This IRand others like it can be downloaded from http://1-1-1-1.net/pages/impulses/index.htm#lorenz195225. The same phenomenonhappens: the reverb gradually turns into a filter, which retains some characteristics of the original IR. Refer to audio examples28 to 39 to hear the result and get the corresponding IRs.

Let’s point out that this method is more empirical than scientific. If we had used another time-stretch algorithm, we wouldhave obtained slightly different results. On a practical note, be warned that very short IRs can’t be used with Audio Ease’sAltiverb 6. This is apparently a software limitation, because there are no problems with Space Designer in Logic, for example.

Other Ways To Modify IR Length

The examples above show that it’s possible to reduce IR lengths with time-stretching. Another way to do that is to usereverb-time modifying controls built into convolution plug-ins. Let’s begin with Altiverb 6, which has two time-related controlslabelled Reverb Time and Room Size.

The latter is the control that corresponds to what we were doing manually,time-stretching the IR itself. It’s supposed to do a better job than a standardtime-stretching algorithm, since it uses an IR-specific algorithm. However, it can’tprovide any drastic IR transformation, being limited to a 50-200 percent stretchratio. That makes reverb-to-filter conversion impossible.

By contrast, Reverb Time doesn’t modify the IR itself, but applies an envelopethat either fades it out before the end, or loops the end of the IR and raises its level,depending on whether you want a shorter or longer reverb.

Equivalent controls can be found in other convolution plug-ins. In Waves’ IR1 plug-in, for instance, the Reverb Time controldoes the same as in Altiverb 6. In Logic’s Space Designer, the Length parameter at the left side of the IR waveform has thesame purpose.

IR Length: Altiverb Vs Peak

Interestingly, the controls built into these convolution reverbs have noticeably different effects from Peak’s time-stretching.Let’s start with the small indoor pool IR we’ve been using to illustrate the continuum between reverb and filtering. Refer toaudio example 40 to listen to it.

Now, let’s compare Altiverb 6’s Time control, set to 50 percent (audio example 41), with the 50 percent time-stretch in BIASPeak (audio example 42). Results are comparable, though Altiverb seems to have removed some low-mid frequencies.

Let’s now put Altiverb’s Time control to 20 percent (audio example 43), and compare it to the 25 percent time-stretch we’vebeen using before (audio example 44). Altiverb’s algorithm provides a much clearer sound, while Peak’s time-stretch soundscloser to the original reverb.

With its Time control set to 1 percent (audio example 45), it appears that Altiverb doesn’t actually reduce the IR length to 1percent of its original state: it’s more comparable with a 12.5 percent time-stretch in Peak (audio example 46). Using Peak’stime-stretch, the walls are perceptually much more present, and the overall result is much more realistic, and much morereminiscent of the original IR. On the other hand, Altiverb gives a much clearer result, which sounds like what one wouldnormally expect from a reverb in a mix situation. Preference may just be a matter of taste. In any case, we’re not going to beable to produce filters from reverbs using Altiverb’s Time control: it’s not drastic enough.

Let’s try the other way around, and set Altiverb’s Time control to 150 percent (audio example 47). This time, Altiverb gives arealistic result: it’s just like the original IR, only longer.

Let’s now switch to Altiverb’s Size control. At 50 percent (audio example 48), the result is very strange, in the sense that thereverb really sounds ‘medium’. This perceptual aspect is so strong it overrides all others. Plus, it definitely doesn’t sound likethe original IR. At this ratio, the Time control appears to be preferable, as does the use of Peak’s time-stretch algorithm.

At 150 percent, by contrast, the Size parameter is much more convincing than when set to 50 percent. Compared to theequivalent setting of the Time control, it’s more musical (thanks to the attenuation of the hiss), but less realistic.

As a conclusion,what’s highlighted is that those controls should be used with caution. They really modify the original IR’stimbre, and don’t stop at manipulating time aspects, as they’re supposed to do. Use of an external time-stretch algorithm suchas Peak’s may be preferable for reducing the IR’s duration.

Diffuse Field Authoring With Noise

As we’ve seen before, a diffuse field is of a continuous and noisy nature. Indeed, any audio content that’s continuous andnoisy can provide a base for a diffuse field, including white noise. Let’s generate white noise with HairerSoft’s Amadeus or anyprogram that can do it. Then, let’s fade it out — a reverb usually fades out with time — and use the resulting audio file as anIR. As heard in audio example 50, it sounds like a very clear, acoustically untreated space. We can do the same experimentwith pink noise (audio example 51): it also sounds like an acoustically untreated space, albeit made from a different material.

Now we want to improve the diffuse fields that are obtained from white or pink noise. What we can do? Filtering the IR willmodify the reverb’s colour, but better still, we can filter the IR dynamically. For instance, we can automate the EQ with whichwe process the IR. Let’s consider how to do that in a way that makes sense. In real, acoustic spaces, when a sound wave isreflected from a surface, its spectral colour gets modified. In other words, it gets EQ’d. Now, remember that diffuse field ismade from a continuous stream of reflections: since the sound is changed each time it’s reflected, it means that the spectralcolour of the diffuse field changes continually.

We can suppose that while the colour itself changes over time, its rate of evolution doesn’t vary: after all, in an actualacoustic space, the properties of the walls will be constant. This means that we could apply a slight EQ at first, which wouldget more drastic over time, while keeping the same overall profile. This would reflect the repeated reflections the originalsound undergoes as it bounces off the surfaces, in effect being EQ’d multiple times using the same settings. Audio examples52 to 54 show the kind of results one can expect from such a technique.

Some convolution plug-ins, such as Waves IR1 or Altiverb 6, include a ‘damping’ parameter that applies a simplified form ofdynamic EQ. Audio examples 55 and 56 show the result of damping in the IR1plug-in. The simple dynamic EQ’ing is clearly audible, and in fact, custom dynamicEQ’ing in your DAW can give much better results. Audio examples 57 and 58 showthe result of static EQ’ing using equivalent settings, clearly illustrating thedifference.

Also keep in mind that it’s not necessary to start with faded white or pink noise:other noises can be a good starting point for diffuse-field authoring, such as tapenoise, amp buzzes, lightly noisy ambiences, and so on. The only limit is your

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controls give surprisingly different resultsfrom time-stretching an impulse responsein a third-party editor.

Waves’ IR1 provides both static EQ anddynamic damping.

0

creativity.

Percussive Samples As IRs

If we look at any given set of impulse responses, we’ll find that a vast majority ofthem feature a decreasing dynamic profile. This seems to be natural enough: if you clap your hands in any given place, theresulting reverberation will not get louder over time. That would be quite absurd. In practice, if you use an impulse responsewith a dynamic profile that is increasing or even stable, you will find that the result quickly becomes incomprehensible. Whilereverse and gated reverbs do, respectively, feature increasing and stable dynamic profiles, their use remains quite specific.

This raises a simple question: if I take any short sample with a decreasing dynamic profile, will that make a suitable impulseresponse? The answer is: that depends. Impulse responses with too much of harmonic content, such as a piano chord, oftenlead to cheesy results. Responses with too many low frequencies, such as a kick-drum sample, can lead to completelyincomprehensible results. Remember that most reverbs exhibit spectra that are quite smooth, without strong formants, andwith quite a lot of high frequencies (canonic models being white and pink noises). When experimenting with impulseresponses, those are good models to keep in mind.

Now, there is one kind of instrument that definitely meets these criteria, and that’s the cymbal. Cymbal samples, especiallyshort ones, can make interesting impulse responses, perfect for eerie vocals or metallic-sounding keyboards. They are alsouseful when you’ve got way too many tracks to fit into a mix, but for some reason you can’t mute any of them (maybe becausethe other musicians or the producer don’t want you to). In this situation, it’s necessary to decrease the ‘timbral largeness’ ofthose tracks in one way or another. Lo-fi plug-ins and filters can help: so, too, can convolution with short cymbal samples, incombination with EQ’ing. Audio examples 59 and 60 are two illustrations of cymbals used as impulse responses.

Speakers & Headphones As IRs

Speakers and headphones naturally change the timbre of any sound that’s played through them, and sometimes add a verysoft and small acoustic ambience, depending on when you put the mic while recording the IR. Generally speaking, cheapspeakers and headphones tend to generate obvious filtering, while the use of more expensive gear will result in more subtlechanges. Such soft EQs can bring interesting results in production, especially when recorded along with the smallest touch ofacoustic ambience — which means putting the microphone perhaps 5cm from the speaker, or 2cm from the headphone driver.Used in mixing, they can bring a distinct yet non-obtrusive acoustic colour to an otherwise dry track. Refer to audio examples61 to 65 to listen to three speaker IRs and two headphone IRs in action.

Impulse responses obtained from speakers and headphones are also highly reconfigurable, meaning they can easilycascaded or combined together. Audio examples 66 and 67 provide two examples of ‘doubled’ speaker and headphone IRs.Other solutions can be found, such as creating coloured echoes from speakers, as shown at http://1-1-1-1.net/pages/impulses/index.htm#SpeakerEchoes.

Space Designer Programs

Digital space designers such as Voxengo’s Impulse Modeler are great for creating special IRs. For instance, you may want tocreate absurdly huge or tiny spaces. As demonstrated in audio examples 68 and69, tiny spaces are particularly interesting for their ability to create highly unusualEQs, with a spectral profile that depends on the simulated surface: cement, gypsumboard, glass, and so on.

Once more, if the effect is not obvious enough during production, cascade it withitself, to create “double IRs”, as shown in audio examples 70 and 71. In any case,don’t hesitate to experiment with different surfaces, as shown at http://1-1-1-1.net/pages/impulses/index.htm#impulsemod.

Finally, learning a little about MATLAB or similar programs opens up thepossibility of synthesizing your own impulse responses from scratch. There’s notspace to cover this topic in print, but you’ll find a guide, with audio examples, atwww.soundonsound.com/sos/sep10/articles/matlab.htm.

Conclusion

I hope this article has shown that close study of convolution brings a whole lot of interesting discoveries: reverbs are in factfilters, tiny spaces produce diffuse fields that can be used as an EQ, cymbals can be used as a reverb, and headphones canbe used as a processing peripheral. To top it all, during IR recording in location, random mic placements can produce better-sounding results than academically ‘correct’ ones (see ‘Creative Impulse Response Recording’ box). Convolution reverbs areopen systems, making impulse responses is easy, so don’t hesitate: forget about realism and correctness, and be creative!

Creative Impulse Response Recording

A well-known and efficient way to create impulse responses is to record them from acoustic spaces. Many articles exist thatexplain the process, including a couple in the SOS archives (April 2005, at www.soundonsound.com/sos/apr05/articles/impulse.htm, and February 2008, at www.soundonsound.com/sos/feb08/articles/logictech_0208.htm). However, it isinteresting to give some thought to the influence of speaker and microphone placement during IR recording sessions. Inconcert rooms, it is usual and logical to put the speakers on the stage, and the microphones in a symmetrical setup in frontof the stage or in the audience. But when recording the impulse response from a living room, for instance, the issue is notso straightforward, and one can legitimately wonder where to put the speakers and the mics.

The first issue is symmetry: in music production, most reverbs are symmetrical. On the other hand, a living room isseldom symmetrical. This means that when trying to use stereo IRs recorded from domestic spaces in music production,one is almost certain to encounter symmetry issues. This is a problem that can be partially solved during recording by usingnear-coincident microphone pairs, such as X/Y setups. This way, the IR will retain some sense of space while beingreasonably symmetrical. Alternatively, it’s possible to agree with Eric James’s opinion in SOS March 2003(www.soundonsound.com/sos/mar03/articles/stereorecording.asp), and consider that symmetry in reverberation is but anaspect of correctness. This means that if it turns out that the IR is not symmetrical, so be it, and any potential problems thatarise during production can be solved during production. This is only reasonable. After all, the main point of a reverb is notto comply with given standards, it’s to bring an interesting timbre. Plus, if we need standard reverbs, we don’t need to makethem ourselves: there are plenty of standard reverbs out there.

Now that we’ve let aside the symmetry issue, we can get more creative. We can envisage an IR recording session in thefollowing way: with one speaker and one mic, we can record lots of mono impulse responses, with the speaker andespecially the microphone being constantly moved around the space. This is not a time-consuming process: whenrecording the IR of an acoustic space, what takes time is to bring the gear to the location, and to set it up. Using thismethod, symmetry issues can be solved afterwards, when recombining those mono-to-mono IRs into mono-to-stereo orstereo-to-stereo ones in post-production.

Let’s illustrate these issues by detailing the recording of the small indoor swimming pool we’ve been using as an examplein this article. This recording was made using two speaker positions and two mic positions, each one featuring a stereo X/Ypair, thus resulting in eight mono-to-mono IRs (see audio examples 72 to 79). When combining those mono-to-mono IRs

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into mono-to-stereo IRs, we certainly can put together IRs from the same X/Y pair, which would be the ‘correct’ way to do it.Alternatively, though, it’s possible to put together IRs coming from different X/Y pairs, which is definitely not ‘correct’, for atleast two reasons: mono-to-stereo IRs created in this way are not symmetrical, and the phase relationship betweenchannels doesn’t compare to anything realistic. Correct and incorrect IRs are represented in audio examples 80 to 83, and,in this author’s opinion, the ‘incorrect’ mono-to-stereo IRs sound much better than the ‘correct’ ones.

Generally speaking, unless you’re recording IRs inside the Amsterdam Concertgebouw, it’s more rewarding to seeksomething that sounds good than something that’s ‘correct’. You can also push this principle further and be creative: putmics near walls, under sheets, under seats, in sinks. Experimentation is the whole point of making IRs yourself. Otherpeople, such as the Audio Ease team, are specialists in discerning correctness and do their job perfectly well. Let themdeal with this aspect of things.

Published in SOS September 2010

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