Monthly Archives: July 2013
If you are looking for another typical subtractive virtual analog synth, you should look elsewhere. If it’s something fresh, innovative and with nearly endless possibilities, read this.
by Rob Mitchell, July 2013
Rayblaster is the latest synth plugin by Tone2. They are mainly known for their synth plugins: Gladiator, Firebird+, ElectraX, and Saurus. If you are looking for another typical subtractive virtual analog synth, you should look elsewhere. However, if your shopping list includes something fresh, innovative and with nearly endless possibilities, read on. Tone2 has created just that, but don’t let me jump ahead too far. Let’s start from the beginning to get an idea of how it all works.
RayBlaster uses a new method for creating sound. Tone2 calls it Impulse Modeling Synthesis (or IMS for short). IMS uses quick bursts of audio, and puts them together to create a unique sound. I have many synth plugins in my collection, and I haven’t heard anything like this before. The closest thing might be a granular type of synth, but those work in a different way. The manual doesn’t state exactly how IMS is accomplished, but from what I’ve heard, it sounds great.
Copy protection is by way of key file – very easy to install, just like many other Tone2 products. Rayblaster is available for PC in 32-bit and 64-bit VSTi, plus 32-bit and 64-bit standalone. For Mac, it’s available in 32-bit and 64-bit VSTi, 32-bit and 64-bit Audiounit and Universal Binary.
For my review, I installed to a PC, and this next part may differ a bit on a Mac. During the install process, folders are created for the different categories of presets that ship with Rayblaster. Those categorized folders end up in a folder called “Rayblaster_sounds”. If you’d like, you can create your own within that and save your own presets there.
Rayblaster has 2 oscillators, and you can load two wav files in each oscillator. You can import your own files too, and in the documentation they describe the recommended way to save your samples for the best results. Rayblaster supports 8 bit, 16 bit and 24 bit wavs in mono or stereo.
The wav file can be a single cycle waveform (they recommend zero crossing at the start), a short sample, or a wave file with the impulse response of a filter.
With RayBlaster, you can also resynthesize a sample you have saved previously, or even a drum loop. You can then sync those to BPM, and you’re able to change much of the sound in real time. There are lots of wav files available with Rayblaster for you to load and experiment with.
The Wave ½ Mix lets you cross-blend between the 2 wav files, giving a kind of wavetable effect. There is also a PulseWidth (PW) sequence, which can be changed over time with the PW value.
The PW sequences and the PW control add or remove certain harmonics from the sound spectrum. I mapped my mod wheel to the PW, and used that while switching between the different sequences available to hear how they differ. You could also map PW to an LFO, or other Matrix source. You can read more about the Mod Matrix later in this review.
Right below the menu to load in a wav for each oscillator, you’ll find the control for the ‘Osc Window’. This can affect the overall shape of the sound for each oscillator, and there are over 20 different types you can choose from. To see/hear what was really going with this section, I checked them out one at a time, just to hear what happens to the sound. The first one called Cosine is good for a basic overall sound, I used that one a lot while changing other parameters. Others like Comb 4X or Sine 8X can really change it up. Together with the Harmonic and Formant controls, there are numerous combinations available.
In addition, you can pick from three different types of noise: Pitch, Formant, and Amplitude; each of which have useful characteristics to add to the sound.
Osc Controls Section
Rayblaster does not have a filter control for its oscillators, but I’ll bet you will not miss it.
You can load any of the included filter impulse responses, or create your own. In combination with the formant and other controls, the audio can be shaped in nearly endless ways. The manual has some tips on how to change some of its settings to emulate eight different filter types.
Formant control adjusts the formant frequency, changing the cutoff of the filter impulse responses and the pitch of granular sounds. It can also just give a bit more brightness to the sound. The High Boost control changes the amount of higher formant frequencies. Using the Damp control, you can soften the sound if it’s too sharp or edgy sounding..
The Harmonic control lets you add more of a dark or bright value to the audio. When negative values are used it will usually be more of darker sound.
Higher values above +1 will repeat the wav, depending on how high you set the value. If you turn it way up, it is nearly inaudible, and can be a little harsh. If you do want it set that way, you can use other controls to tame it a bit, such as damping or using a different type of Osc Window/PW setting. Normally you won’t be cranking that way up into that range, but it is available.
The Start value is the start offset for the waveform you’ve loaded up in whichever oscillator. You can get really creative with this, as the Start amount can be used for granular or looped type of sounds. It can play forward or backward through a sample, depending on how the Start value is set. Playing around with this feature was one of the coolest parts of Rayblaster. You really must check it out.
Sync, Ring, Phase, and Drift are all included. They’re handy controls that Tone2 has added, and can help with getting some classic synth sounds, among others. If you set Phase to 0, the oscillators will be always running. If it’s set to a value above 0, it will use that same phase value you set it to, and re-trigger both oscillators every time a key is hit.
Modulation, FX, and More
There are three Envelopes included. One is set to only be used with the Amp section. The other two can be assigned nearly any way you want via the Modulation Matrix.
Using the oscillator controls in conjunction with the Modulation Matrix can give some dynamic results. Match up the Formant control to an envelope, set Wave Mix to another LFO or the Mod Wheel, and you’ll start to get my point. The Matrix has three sections with five source/destinations available on each of the sections. This provides plenty of room for expanding the sound. You can get to some added goodies in the Mod Matrix as well. For example, besides the three types of noise in the Osc section, of the synth’s main screen, from the Mod Matrix you can also get access to white and/or pink noise. For a full list of sources and destinations, see the figures at the end of this review.
The Value Modifiers in the Matrix add even more flexibility. There are four different modifiers to choose from, and they can give a different type of control over whatever source you have assigned to it.
You also get two LFOs, each with six different waveforms available. These can both be assigned in the Mod Matrix.
The 16 step Arp/Gate is very flexible, but it could be little easier to use. But once I got used to it and all its options, I really liked it. There are also many pre-made Arp patterns you can load, and you save your own for future use in other presets.
There are two effects available per preset. You can pick from reverb, delay, distortion, phaser, flanger, EQ, and more. The presets are high quality and very useful. I really liked the multi-tap delay, trance gate, and reverbs.
A couple of handy menu items included are the ability to copy Osc1 to Osc2, and exporting of wavs from the oscillators to disk. You can also initialize just the Matrix or Arp, while keeping all the other settings.
One feature some will appreciate (but it has nothing to do with sound) is being able to hide the keyboard. Some people just don’t like keyboards on their synth plugins. I don’t mind them myself actually, but it’s nice to have this option. Some may use Rayblaster on a smaller laptop screen for instance, and this would help save on the screen real estate.
With Rayblaster, I found that I was having loads of fun experimenting: changing the harmonic controls, formant knob, and using the Arp to control many aspects of the sound. That’s only part of its sound. It can also be set to more “standard” kinds of sounds, imitating other classic synths if you’d like. The manual gives some examples of ways to attain some of those types of sounds. Tone2 also gives you tutorial presets where they used those methods that are mentioned in the manual, and you can check them out to see how it’s really done.
One thing I’d like to see is the effects available as destinations in the Mod Matrix. Also, and I might be greedy here, but I would like three effects available. Two is OK, but just one more would really round-out the effects. I usually like delay and reverb much of the time, but then I can’t add an EQ, chorus, or anything else.
I thought it would also be useful if you could save the Matrix settings, so then they could be loaded into another preset. I found myself repeatedly setting up certain functions for different presets I made in nearly the same way each time: LFO amount to Modwheel, OscFine to the Pitchwheel, or Formant to Modwheel, etc.
So is IMS the next big thing? However IMS works, it sounds very good, and the synth controls make it extremely versatile. Rayblaster is a far cry from most other synths on the market. It lets you get the older classic sounds, if needed, while also breaking new ground itself. It would be nice to know exactly how IMS works “under the hood”. It is explained a bit more on their website. But not understanding the underlying technology is not a big deal…I’m having too much fun just working with this great synth.
At 199USD, Rayblaster isn’t exactly cheap (nor are the two available expansion sound sets). But one thing’s for certain: you don’t already have something quite like it in your instrument arsenal, so it definitely may be worth a look/listen.
Effect aficionados (would that be “efficionados”?) will be in seventh heaven with this collection of VST plug-in that cover the full range of sound sculpting capabilites.
by David Baer, July 2013
Deal of the Century?
In March of this year, on-line vendor Don’t Crack offered a group-buy for a 40-piece ensemble of plug-ins called the Plug & Mix V.I.P. Bundle. The bundle at that time retailed for $399 and the group buy target price was $99 when sufficient subscribers were signed up. Not being one who can resist a $2.50/plug-in offer, especially for plug-ins sporting rather delightful user interfaces, I jumped on the bandwagon irrespective of not having downloaded them to demo first. The target price was achieved, and thus I’m able to offer a review of this package today.
First let’s get the basics out of the way. The bundle is available for all current mainstream formats on PC and Mac in both 32-bit and 64-bit. A customer-friendly authorization system is all that is required and the demo restrictions allow plenty of freedom to take them for a thorough test drive. The possibility exists for additional effects to be added to the bundle in the future, and customers who own the bundle are promised free upgrades.
At the time this is being written, the bundle sells for $249 (on sale from $299), and individual plug-ins can be had for $39. I believe that in the past, Don’t Crack had a deal that offered a free upgrade to the full bundle after some number of individual units was purchased, but that doesn’t seem in evidence at the moment. In any case, as of right now, the per-effect price is about $6.25 … not as good as the group buy opportunity but still not bad if there are a sufficient number of plug-ins that you’d actually use.
We’re not going to look at every plug-in here — not even close to all of them. Instead we’ll take a look at a handful close up to hopefully give you enough of an idea of what to expect in the overall collection. Before doing that, though, we can consider some common characteristics shared by most.
First, they are all simple, with very few knobs and a user interface that’s so intuitive you’re not likely to need documentation. Even where function isn’t evident, there are so few controls that a little experimentation is all that’s needed to understand how to use any of these devices. All of the units come with factory presets and the ability to save user presets. While useful, the effects have such simple control panels that presets don’t buy all that much.
While the UI design aesthetic is of no consequence as far as the final sound, I have to say the V.I.P. “front panels” are delightful and can bring a little joy into an otherwise dreary day. Visually, they rock.
Their functions are all over the map: distortion (amp simulation, bit reduction, etc.), compression, EQ, reverb, and so on. All the conventional bases are covered. Most of what you get will duplicate functionality you probably already own if you have any mainstream DAW installed. So why bother considering the collection in the first place? Here’s the reason: You will probably find a handful of effects that you consider real gems, even if you think the majority are unneeded in your setup. If the price is right, you may find the bundle a worthwhile purchase. The hitch is that everyone is likely to have their own list of favorites, so you’ll have to work out on your own which ones the gems are.
Another quality that makes this bundle attractive is that the effects beg to be assembled in chains. Again, their simplicity and focused purpose make them not only easy to use individually but also in groups. If your DAW supports effect chains (e.g., as does SONAR), you might have a ball inventing your own custom composite effects.
With that out of the way, let’s look at several of the plug-ins close up. I selected the first two of these because for me they’re the “pick of the litter.” The others were picked just because I think they are nicely representative of the group as a whole.
I fell in love with Dimension 3D at first listen. It is said to be based on the Roland SDD320 “Dimension D” manufactured in the late 70s. Don’t Crack asserts that the unit “adds the highly sought after detuning effect heard on countless classic rock records and continues to be used on modern club tracks.” The problem with that claim is that if the effect is in evidence to the point it’s noticeable on a recording, it may not be a sound you want to embrace. As with so many plug-ins in this collection, a little goes a long way. It doesn’t take much to overdo it with Dimension 3D, turning gorgeous warmth into cloying suffocation. But at more restrained settings, the transformations preformed can be gorgeous.
Other than the input trim and output gain knobs, there is just one knob: Detune. Along with this we have Mode selectable via three buttons: D1, D2 and D3 (D for depth amount perhaps?). Documentation? We don’t need no stinkin’ documentation! You can find the setting you like best through experimentation faster than you could open a help file in the first place.
The Dimension 3D is not the only chorus in the bundle, by the way. We also have a more conventional chorus effect available in the P&M Chorus Ensemble.
My other favorite is the Cool-Vibe, said to be based on the Uni-Vibe, a stomp box from the late 1960s used by artists from Jimmy Hendrix to Pink Floyd. According to Wikipedia, the Uni-Vibe, “though often associated with chorus, is in fact created through a staggered series of phasing filters, unlike the usually aligned filters of a normal phasing effect.”
Like the Dimension 3D, the effect can all too easily be overdone. With restraint, this effect can turn something merely pretty into something absolutely beautiful. But turn the dial up just a bit more and you’ll think about opening the windows for some fresh air.
You can control the speed of the effect — it does provide right-left image movement and the speed can be synced to host temp. The width of movement is controlled by the Width knob. Three modes are offered: Sweet, Mellow and Deep. This strikes me a little like asking “do you want that small, medium or orange?” But, no matter, like all the other effects in the collection, the simple interface makes it easy to rapidly find a usable setting with only brief experimentation.
The bundle also offers an auto-pan module called Tremolo Pan which does what it says and is very basic. Cool-Vibe can impart some of that quality as well, but can produce so much more of a beguiling quality in my opinion.
Next we look at one of the more unique offerings, the Granulizer. Native Instrument’s Absynth synthesizer has a stunning effect called Aetherizer. It chops audio into short “grains”, manipulates them in various ways and recombines them to produce completely transformed (or just partially transformed) sounds. Absynth can be used as just an effects unit to process the output of other synths. However, it requires MIDI note events even in this capacity. Sending MIDI to an effect can be somewhat of a pain to set up in some DAWs. So, I had high hopes for Granulizer because one might expect it will do largely the same without the MIDI routing hassle.
However, I was disappointed in the inability to reproduce the quality of the Aetherizer effect. It appears that Aetherizer uses the MIDI note events to do more than just open gates. While Aetherizer can produce some ethereal transformations, Granulizer’s agenda tends towards chaos.
So what does Granulizer do to sound? It … um … “granualizes” it? This is one of those things that you just have to hear — it’s too difficult to put into words. Trying out the presets should get you well down the path to understanding this effect as long as you start with simple sustained sounds so that you can get a sense of what it’s all about. That’s not to say you might not find it of use with non-sustained sounds. But it can totally annihilate the identity of the original sound, so when you evaluate it, feed it something simple and easily recognizable to get a sense of how Granulizer rips sounds apart and reassembles them.
The Leslie Mystique
I’ve never played an organ with a Leslie rotating speaker, so I cannot be a great judge how well the LS-Rotator effect pulls off its Leslie simulation. But although I haven’t performed with one, I certainly have heard them, and I must say that this one sounds like the real deal to my ears. I think it’s absolutely stunning. Try it on electronic organ, by all means, but don’t stop there. It does magical things with synth strings, for example. One of the great things about this effect is that it’s a plug-in. I have Leslie effects in several synths, but they are on-board effects. With LS-Rotator, I can place a Leslie speaker simulation on virtually anything I like. Better yet, this effect has more control options than any synth-resident Leslie effect I’ve seen. For example, there’s a Spread slider control that controls the width of the stereo image.
Unlike many of the other effects in the bundle, this one is not so easily misused by over-driving it. At high drive settings, the results are gritty but delicious to my ears. A quick trip through the half-dozen or so presets will give you an excellent demo of what to expect. The LS-Rotator is a total winner in my book. My only regret is that I can’t completely turn off the speaker-rotation effect and just use the distortion processing, which I have a feeling might sound righteous on its own.
Pass the Dust, Please
One of the more unusual effects in the V.I.P. Bundle produces vinyl record surface sounds. Three types of sounds can be combined: background surface noise, dust and scratches. Controls allow you to set surface noise amount and tone quality and set dust and scratches amounts and rates (platter rotation speed) individually.
There are four factory presets with this one. One has the amusing name “Dirty Needle.” Another name, one I hope the developers wish they could take back, is “Amy Crackhouse.” Tsk, tsk!
Admittedly, this is not the sort of thing you’re going to be reaching for very often. But it’s nice that the bundle does have some reasonably unique effects in its bag of tricks. There’s not a lot more to say about Vinylizer. It only does something that’s well off the beaten track, but it does it very well indeed.
VIP – Very Interesting Purchase
For $99, I couldn’t be happier with my purchase of the V.I.P. bundle. Of the 40 effects, there are some that I am unlikely to ever want to use under any circumstances. Then there are the mainstream mixing plug-ins that duplicate functionality I’ve already got in spades: compressors, EQs, limiters, etc. What’s left is a collection of plug-ins I’m delighted to have acquired.
Is the V.I.P. bundle for you? Well, first of all, if it ever comes up for sale at $99 again, I’d suggest that you just jump on it. At the current price of $249 it’s obviously a more difficult call. Basically, you simply need to download the demo package, find which effects float your boat and determine if there are sufficient ones to warrant your purchase of the bundle. You probably will come up with a three-part list much as I did: effects that you flat out would never use, effects that you like but that duplicate the function of things you already own and like at least as well, and finally, effects that won’t want to be without. If that final group is too short of a list to justify buying the bundle, you still have the option of picking up them up individually for $39.
More information here:
New Sonic Arts have released a second offering next to their excellent granular sampler Granite. Please welcome the software sampler Nuance to the world.
by Robert Halvarsson, July 2013
A No-Frills Sampler for the Future
Granite was an extremely interesting concept and a granular sampler which this reviewer has used in more than one of his projects. Innovatively, it used the GPU to offload work form the central processor(s), thereby making it more effective to use with modern computers.
Nuance pretty much builds on the successful and tasteful design choices of Granite, marketing it as a “no bloat” sampler, an alternative to its feature ridden counterparts. Nuance ships with very few instruments, but the main idea is for the user to dig out their own sample library and make Nuance twist things around.
It supports the standard fare in waveforms, you can load .wav and .au-files and soundfont-files (sfz). Inside the sampler, you get LFOs, analog modelled low-pass, band-pass and hi-pass filters, simplistic compression and overdrive controls. The effects are quite vanilla, but at the same time they do what they should. Nuance gets rid of a lot of what may be unnecessary when working with samplers, thereby letting you focus on the core of things.
You can, of course, map samples out to your hearts delight, layer them via a piano-roll, and get funky with things, but you can also modulate the effects using a nifty and flexible routing system. The modular possibilities are very nice, and quite easy to get into as well, which results in what may be considered the most original aspect of Nuance. You also get various loop modes and what could pretty much be considered unlimited polyphony (128 to be specific).
Knowing that New Sonic Arts still supports Granite with free updates, people who take Nuance to heart most likely will have something to look forward to. When writing this article, the attack, decay and release ‘time’-control (ADR) just became available as a modulation targets. Not bad, not bad at all… And seeing that a form of well-articulated simplicity has been a growing trend (think of the success of Apple to illustrate this case in point), it is likely that Nuance could appeal to quite a large demographic of producers. After all, it’s all about workflow these days.
So be sure to check Nuance out, and while you’re at it, try Granite out if you haven’t. It has been my gateway to granular synthesis – and certainly deserves more recognition then it has gotten. As with Granite I believe we are in for a treat when it comes to the development of Nuance as well.
New Sonic Arts
Hideaway Studio is becoming a far cry from a hidden treasure. We look at two superb offerings from this relatively newcomer in the Kontakt sound offering marketplace.
by David Baer, July 2013
Hideaway Studio (http://hideawaystudio.wordpress.com/) is a recent entrant into the Kontakt-sounds vendor space. Since their first offering in Dec. 2012, they have been producing new sound sets at a prodigious rate … making some people wonder “where the hell did these people come from all of a sudden?” Well, first of all, Hideaway Studio is just one person: Dan Wilson. Prior to Hideaway Studio becoming a must-have-bookmarked Kontakt sound supplier, Dan was probably most widely known for his restoration of 1938 Novachord. That story makes for fascinating reading if you’re into ground-breaking early electronica: http://www.novachord.co.uk/.
Dan then collaborated with Hollow Sun to produce two Novachord sound sets for Kontakt. It was this collaboration that set the wheels turning that would eventually get Hideaway Studio on track to be the preeminent sound factory it has so recently has become. Dan is an electronics wizard and by some accounts a bit of a mad scientist (Mr. Howell, no worries … I have no intention of revealing the source of that information!). Fortunately, his interests seem to focus on vintage sound gear of all sorts … to our very great benefit. The result is the growing collection of wondrous, unique and fascinating sounds for Kontakt being made available from Hideaway, all for remarkably low prices.
In this article we’re going to look at Hideaway’s first two major offerings, both released in Dec. of 2012: The S-VX Hybrid Library and The Orbitone Collection. Both of these require the full version of Kontakt, as will most of the packages that will be reviewed in this column in the future. For the most part, we’ll be looking at attractively priced offerings in this column, and the only way sound producers can keep the prices low is to avoid the royalty payments required by NI for free Kontakt player compatibility. If you do not own Kontakt, just be aware of the fact that every month brings new and compelling reasons why you should consider taking the plunge and acquiring it.
Combining S-VX Hybrid Library and The Orbitone Collection into a single review makes sense for several reasons. Most significantly is that they offer very similar capabilities. They are essentially the same Kontakt instrument delivered with different underlying sample sets. Hideaway asserts the two are quite complementary when combined … an assessment I’m very inclined to agree with.
We’ll start by examining the S-VX library (hereafter, “S-VX” for brevity). By the time we get to the Orbitone collection, all that needs discussion will be the samples. If this instrument looks like something you might get from Hollow Sun, it’s no coincidence. Hollow Sun’s Stephen Howell did the interface design and Kontakt scriptmeister Mario Krušelj did the layering engine script.
The sample sets for S-VX share a common sampling process. Hideaway got its hands on an S900 sampler (pictured) which was manufactured in the mid-80s. This was a 12-bit machine with a whopping 750 KB of memory and used a floppy drive for sample storage. But the story doesn’t end there. In Hideaway’s own words:
One feature of the S900 that was often overlooked was a curious 13 pin connector on the rear panel marked “VOICE OUT”. … AKAI had released a small number of polyphonic analog synthesizers during the same era as the S900. These included the AX73 keyboard and the VX90 rack mount. They have often been overlooked over the years possibly due to their rather control-free front panels and menu driven user interfaces. … The infamous 13 pin connector permitted the sampler to be used as a complex oscillator source driving the filter stages and VCAs modulated by the envelope generators. In short, you get to mix 12-bit digital crunch with the warmth of analog filters!
When you first encounter the S-VX and play through the presets, you’ll hear a collection of lush, rich-sounding, multi-sample stacks of sound. But turn off all the effects and three of the four layers and listen to individual samples in isolation. These sounds are so thoroughly vintage that you may feel the urge to sneeze at all the imagined dust.
The 32 different sounds (enumerated in the reproduced menu at the end of this review) are great by themselves, all being expertly sampled as we’ve grown to expect from Hideaway. But their real strengths become evident when layered, either slightly detuned (and perhaps pitch modulated) stacks of the same sound, or in complementary combinations. As can be seen from the user interface, building these layers is trivial, and I can think of few instruments (Kontakt or otherwise) where the user should feel so motivated to do some preset design of their own.
Each of the four layers has identical controls:
- On/off, level and pan position
- Envelope (attack and release, but no decay and sustain)
- LFO (directly wired to pitch) rate and depth (bi-polar so you can get layers to move opposite each other if desired)
- Tuning: semitones (more octaves than you’re likely to need) and fine tune
- Tone: an EQ curve control that is hard to briefly explain but the following image tells the whole story.
The four layers share an effects chain consisting of a reverb, an echo/delay unit, a phaser and a chorus. The reverb uses Kontakt’s excellent convolution engine and comes with 24 supplied types, pictured at the end of this article. The effects all sound great if used with restraint. But they are also the source of the only negative comment I have about S-VX. The presets suffer from Omnishpere-itis, being sound so drenched in verb, delay, etc. that they’d rarely be of use in a real mix (although they sound lush for solo demoing). But this is a trivial complaint as it takes no effort to adjust these seasonings to taste.
Let’s now turn our attention to the sibling offering, The Orbitone Collection. As said earlier, the layout and capabilities are identical to that of the S-VX, as you can see in the graphic below. What are different are the included samples … in this case 28 (again, listed at the end of this review). A whole battery of devices were used in their capture. In Hideaway’s own words:
1972 Eminent 310U (strings, pads, resonator choir, e-piano, organ), 1976 Minimoog (brass, pads, bells, chimes), 1938 Novachord (strings, e-piano), rehoused 1978 Polymoog Formant Resonator Section (choir), 1976 Revox G36 tube half track tape machine (g36 choir), Panoramic tube Dual Tone Generator (chimes, e-piano), two 1967 Heathkit EUW-27 tube signal generators (chimes, e-piano), ARP Omni Chorus section (strings), Hideaway Studio Triple Tube Hybrid Phaser (evolving pads, phased chimes) and Dual Tube Hybrid Filter (underwurlde sweep), Tube Ring Modulator (bell ratios), Discrete Dual Exponential Sawtooth Generator (french horns), Tube Overdrive and Passive Triple L/C Resonator buffered with Y-amplifiers from a 1968 Tektronix tube scope! (deep resonator vox), All sound sources captured via two Hideaway Studio Type TEQ-9B Active Tube EQs in 24-bits with the RME Fireface.
For all of that, the star attraction of the show is the Eminent 310U, a hybrid organ/synth beast from 1972 that Dan rescued and restored with the help of a fellow named Albert Steenbergen, who reportedly kept these instruments in working condition for notable 310U enthusiast Jean Michel Jarre.
More on the 310U, again in Hideaway’s own words:
The 310U boasts really quite an unusual architecture permitting a mixture of sustained and percussive envelopes to be applied to combinations of all of the timbres which can be layered together. There is also a gorgeous swirly six-stage analog stereo chorus “Orbitone” section and built in spring reverb. The instrument was significantly more complex than most combo organs of the era due to in the main to the polyphonic percussive and sustain controls using discrete analog technology throughout.
And so now we know where the Orbitone name comes from.
Orbitone and S-VX share much in quality of sound, offering a broad range of lush, complex layered sounds. It’s hard to imagine that anyone would be attracted to one of these not being attracted to the other. Furthermore, they complement each other beautifully. At 10 pounds (or roughly 14 USD) apiece, these libraries are a marvelous value. On the other hand, maybe they should be viewed with the caution one would a “gateway drug.” Once you get the Hideaway “habit,” resistance purchasing all their offerings may very well be futile.
Would Beethoven been a better composer if he had access to Liquid Notes? Probably not … but mere mortals like you and I, that may be another story altogether.
by A. Arsov, July 2013
Back to the Future
In the past, I have done mostly electro IDM music. Whenever I tell people that I’m not a live musician, most of them imagine that I’m sitting in front of my computer, whispering to the monitor what to do, while my computer makes music instead of me. The rest of the world makes a decent living with their hands, while I’m faking to be a musician stealing jobs from real musicians.
And know what? Suddenly all those nightmares of the honest working heroes become a reality. Liquid Notes is a boogeyman that comes out of the computer to steal our brains, taking control of our lives. Run working hero, run… Stefan, Stefan, Roland, Karl and Gerrit will eat you alive!
OK, not really, but yes, they make it happen. They make a program that has some sort of artificial intelligence and – it works.
Liquid Notes is a “production tool that assists you with chords, scales, and harmonic movement with ease and efficiency.” That is a description from the Liquid Notes home page, and they are right. You feed the software with some simple MIDI arrangement, it will help you to develop that idea much further by changing harmonies, adding new ones without screwing the melody and making Stockhausen out of Bach. And talking about boogeyman, the aforementioned boogeyman team are one of the most supportive groups of people I have encountered in the music industry. We talked a lot even before they figured out that I had gotten their software for reviewing purpose. (I obtained it from Bestservice.de) They helped me to sort some issues even when they thought that I was using the demo version of their product. Their chief developer Stefan also offered me his mobile number while we tried to fix a problem with the virtual MIDI cable.
The software is fairly simple (and they promise that it will be even simpler in the future). You can start with some simple arrangement, a few bars filled with chords, bass and lead line, and export that as a MIDI file. The truth is that you can also use any other MIDI file, from any other author, changing it with the software and making it fairly unrecognizable. This can be ideal for remixing anything. After you have that MIDI file, you just need to open Liquid Notes, importing that MIDI, selecting the rank of every MIDI track, letting the program know if that track contains melody or chords.
More or less that’s all there is to it.
Open your sequencer with that MIDI and start tweaking Liquid Notes. Of course it is not an almighty program, so if you push those sliders too far from the origin, the results can be closer to Stockhausen than to Bach after all. But use it carefully and you will be more than happy with the results. Setting it up for the first time is a bit tricky, but as soon as you’re in the saddle everything becomes easier. I spoke a lot about work-flow with Boogeyman team, and they are aware of the complexity of the whole process. OK, it could be that I’m also a bit spoiled, my DAW is a bit slow, so all that open, close, open routine is a bit time consuming. They are working hard trying to find a solution, but no matter how many times you open your DAW (to make a simple arrangement), to close it (as Liquid Notes should be opened before the DAW) and to open it again, this program offers such rewarding results that we can easily forgive all those minor difficulties.
I’m aware that the simpler a program looks, the more complex things may be under the hood. So all you need to do is to watch the few video clips they have on their site, clicking a few knobs, dragging a few sliders and you’ll be a new Bach, or at least a more advanced you. Visit Bestservice.de, find Liquid Notes, and pay 159 EUR or 175 USD to feed the Boogeyman team. Maybe you are not aware about the fact that Stefan, Stefan, Roland, Karl and Gerrit are real people who spent several years working hard with their own hands to make this happen. Now it is your turn to whisper in the monitor!
Interview with Stefan Lattner – Project Manager and Chief Developer
SB: You are not just a programmer and hobbyist musician, as Liquid Notes proves. You clearly have excellent knowledge of harmony. Can you tell us a bit more about your musical background?
SL: Yes, I do have quite a deep knowledge of harmony and music theory, but this actually wouldn’t have been true if you had asked me before I started work on Liquid Notes. Certainly, I have picked up bits and pieces of music theory during my musical life but I could achieve all the tasks needed for my work by using my ears and by being gifted with a quite decent feeling for music.
I studied the violin and I am an autodidact in playing the piano, the guitar, and in song writing. Additionally, I went to a program of media technology in college where I achieved audio mixing and signal processing skills.
I love composing songs for my music group, “Mariachis de las fiestas locas”. And I love live staging as well as mixing our recordings. Currently, I am writing my Master’s thesis. The goal is to teach the computer in music composition by having it listen to examples.
All of that has led to a certain understanding of music theory, but most of the time I had to rely on “trial and listening”. That is an approach many musicians choose and that’s the philosophy behind Liquid Notes, too.
SB: Regarding harmony, I presume that you, along with the other Stefan and Gerrit are working together on that issue. Can you tell us, who does what?
SL: The other Stefan is our visionary, he chiefly deals with the perception of music and its effect on listeners. And nothing goes public without his approval. However, all the harmony-related models where formulated by Gerrit. I assisted him by transforming his theoretical models in a form the computer is able to understand. During this work with Gerrit, I gained my harmonic skills mentioned above.
SB: You may be bringing into reality the biggest nightmare of all purists – you have made software that almost makes music like a human – some sort of artificial musical brain. Where do you think is the strongest point of your software, to help us to compose or to help us to find harmonies that we maybe didn’t think to use – to surprise ourselves?
SL: Consider a few seconds of any musical piece. There are several properties one could recognise: timbre, tempo, scale, harmony, rhythm, several voices, genre, contour, and so on. We look at music as an interplay of changing and recurring properties and how they affect human perception. Putting it that way, we offer a tool for easily changing one of several properties to enhance the psychological effect on the listener. Therefore, all would be true: helping to find novel harmonies is one way to help composers. Enabling our users to switch through different harmonies effortlessly is a way to optimize the harmonic properties of a song. And so it is also a way to improve the quality and speed of the composition process.
SB: The program looks fairly simple, that probably means that you spent enormous time and energy to implement all the rules and interactions that are crunching somewhere in the background. What is the hardest nut you had to crack during the planning and/or programming process?
SL: You are right, Liquid Notes incorporates a hidden process chain which users might not be aware of. If one link of that chain fails, the entire process of harmonic substitution, including the recommendation feature, would fail. That chain consists of a track class detection, meaning that it classifies instrumental tracks into bass, melody, chords, etc. With the actual harmonic analysis, we then identify scales and chords in the arrangement. The recommendation table for chords connects, sorts, and organizes chords. It takes into account the previous chord and the actual arrangement of chord notes (e.g., inversions).
One of the most challenging links in that chain was definitely the harmonic analysis because we have to cope with any type of arrangement the user opens in Liquid Notes and therefore this module has to be very robust against “compositional peculiarities”. Another one was the reharmonisation module because there are countless possibilities to model a certain harmonic structure. But also the recommendation table was a hard nut to crack and we are constantly improving all those parts taking into account all the feedback we get from our users.
SB: Your future plans? What do you think that can be still improved in Liquid Notes?
SL: We are currently working on improving the usability and the coherence of Liquid Notes. It’s hard to find the right tradeoff between simplicity and providing the user with the desired information. For the next releases, we plan to include an overview bar to make the overall navigation within a session easier. Additionally, we are checking on ideas about how to display chords in different ways. Soon, we will extend the list of sequencers to which Liquid Notes can then connect automatically. There are still a few tutorial videos to do in our pipeline and also our webpage will get redesigned.
Finding a certain chord instead of having to stumble through the entire set of recommendations will be another necessary improvement to add. And there are virtually hundreds of little improvements waiting in our pipeline to turn Liquid Notes into a yet more mature product.
SB: Imagine that a Hollywood director wants to know more about Liquid Notes, so, we need a synopsis – just a two sentences that will tell us everything that we should know about your program (and to convince us to buy it.). How would you respond?
SL: Imagine that director and his composer sitting in the studio together listening to the new score for a film for the first time. And let’s suppose we are dealing with a digital score here. Under traditional circumstances, this is a situation full of suspense because a negative reaction by the director or the producers could mean that the score has to be redone from scratch. Unfortunately, as we know, directors and producers are always in the dominant position.
With Liquid Notes, the composer could simply push a few buttons and add a few harmonic twists here and there, changing the emotional effect of the score entirely. Not only could this safe the situation but even if adjustments to the score had to be made, the direction would be perfectly clear now. The composer can present several variations with ease, and he can do it instantly.
SB: I used to beta test virtual instruments and effects, always finding a way to crash the tool I beta tested, but my knowledge of harmony is not good enough to shoot down Liquid Notes. I wonder if here is any weak link harmony-wise that you intend to improve in the future?
SL: Not really, no. 😉 But truth be told, Liquid Notes in its current state is not a product to tackle complex counterpoint or similar intertwined harmonic structures with. But let’s wait and see …
The Fab Filter duo have done it again, this time with a superb tool to combat dasssstardly, disssscusting ssssibilance in your vocal tracks.
by Robert Halvarsson, July 2013
Sss Stands for Sibilance
FabFilter has been one of my favorite virtual effect companies for several years now. Why? Because they make products that are both visually appealing and well sounding. With Pro-DS, they have set out to create a one-stop solution for de-essing work.
It is considered one of the harder things to accomplish in the virtual world – a De-esser plugin which at more than moderate levels doesn’t destroy the original voice or which mistakes undesired sounds with hyped vocals with high frequency sheen. In order to accomplish this difficult task, FabFilter has written a new, custom algorithm, thereby taking the long road in attempting to tame high and obtrusive voices, instead of resorting to tried methods of the past.
The goal is to create a transparent de-esser. And to go with that, we get all the usual good stuff that we’ve come to associate with FabFilter for the last couple of years. Amongst its features and especially worthy of mention is the interactive MIDI Learn, internal undo/redo function, sample accurate automation, and last but not least, the outstanding interactive help function. And don’t you worry, you pro’s out there, you can turn the help off and wander aimlessly in the dark, if you want to.
One of the standard processes that makes a de-esser works is having a compressor that targets certain frequencies in the high register. The goal is for the voice to be remedied of its annoying aspects. The only problem is, this process is usually not intelligent enough to distinguish between desirable parts of the vocal in the higher regions, and things it should leave alone.
In other words, you will potentially not only push the “ess” back in the mix, but also other parts of the vocal as well.
What Sets Pro DS Apart
The unique newly developed Single Vocal algorithm in Pro-DS has an uncanny ability to separate sibilance from non-sibilant material in individual vocal recordings, even though it may be part of the same frequency spectrum. The idea is precisely this: to let you lower the “esses”, but not punish the other parts of the vocals.
But it’s not only about the vocals. You could use Pro-DS to tame high frequencies in other instruments as well, or entire mixes. When using it in this way, it works like a multi-frequency band compressor, but instead of a true multi-band compressor it only targeted high frequencies. Since this is what it is tailored for, it can be more effective and also, more transparent in these cases.
For those ready to go deeper into Pro DS territory, a great way to begin is to wander through the few but great presets. You have male and female vocals, split and multiband presets, and also presets for whole mixes. There are not a whole lot, but there needn’t be.
For those who want to create a preset from scratch, you can start by navigating the two complementary modes, single vocal and general mode. You also have wide band and split band options, which is perhaps a bit trickier to understand. Without going in to much detail, wide band works aimed at high frequencies, but when lowering the impact of the undesired sounds it will also target the entirety of the frequencies in the full spectrum. This is considered to go best with single vocals – split band will add some latency and leave lower frequencies alone, thereby being suitable and more in line with full mix work or more complex vocal material. By putting down your area of focus, variable between 2KHz to the limit of 20KHz, you will focus on the threshold of the material and the range, which together will achieve how hard Pro-DS will work on your material.
De-essers or de-essing is usually not the first thing a music hobbyist would turn to when working with vocals. Besides equalizing, people usually apply general compression and some reverb. But a dedicated de-esser can be a good add-on for a purposeful vocal chain, and downright essential when working with vocals with strong sibilance.
FabFilter Pro-DS is a very strong candidate for this type of dedicated work, because it is tailored for this specific task alone. While remaining quite easy to handle, it can deliver smooth, great-sounding results while retaining much of the original sound, making it suitable for almost surgically precise work. It has an almost organic quality to it, which is quite rare. Therefore, it is warmly recommended, and should be thoroughly examined by those who value quality vocals.
Price and Availability
FabFilter Pro-DS is available for EUR 149, USD 189 or GBP 124, supporting both Windows and Mac OS X in VST and VST 3, Audio Units, AAX, RTAS and AudioSuite plug-in formats. Bundles with FabFilter Pro-DS and other FabFilter plug-ins are also available. See www.FabFilter.com/shop.
We take a close look at a highly capable software synth that would be impressive at twice its price. Synthmaster seems worthy of its name.
by Rob Mitchell, July 2013
SynthMaster by KV331Audio has been getting some great press lately, even reaching #3 in a recent Computer Music reader poll. Some describe it as an all-in-one type of synth, and as you’ll read in this review, it seems they are right. Is it worth all the hype? Let’s dig in deeper, and check it out in a bit more detail.
SynthMaster is a two-layer semi-modular synth, each layer offering two main oscillators, multiple filter types, and a huge set of additional features for modulation. It is available in 32/64-bit, VST/RTAS for PCs, and VST/AU /RTAS formats for Mac OSX. The Factory version has over 750 presets, and there are many optional preset banks for it.
From my own experience with SynthMaster, I can say KV331Audio didn’t pull any punches with their monster synth. It is tough to find something it can’t do, especially modulation-wise: It boasts over 95 modulation sources and a whopping 650 mod targets.
In each of the two layers, you have your choice of a standard Oscillator setup, or you can use an Additive, Wavescan, or Vector oscillator. There is an “Audio In” oscillator you can use to route your own audio through.
Voice-wise, you can choose from Poly, Mono or Legato, and the overall poly count can be set to use up to 64 voices. You also get a Unison to use in your preset design, and it can go up to 8 voices. If you want to change the ways the filters work, you can choose from Split, Parallel, or Series. More on the filters later.
With the standard Oscillator, you get typical waveforms such as Sine, Triangle, Square, Sawtooth, Pulse, and Noise. Plus you can pick from tons of other single-cycle waveforms to load in place of those, or you can load up an SFZ file. It’s easy to load them; you just click the left-right arrows to cycle through the waveforms. You can also right-click on the picture of the waveform, and it will bring up a menu to get at exactly what you want. From that same menu, you can also load in WAV/AIFF files as an SFZ. After importing a wav (or wavs) and it is converted, it will then show up under SFZ Instruments, in the UserSamples folder. You can also drag/drop wav(s) on to the oscillator itself.
With the Additive setting, you can load a different waveform for each of the eight “simple” oscillators. They’re called simple, but they’re still oscillators. They just don’t have some of the extras of the standard oscillators. You can change the volume, pan, detune, and modify frequency of each one of those oscillators. If you set all four as Additive (two additive oscillators on each of the two layers), SynthMaster can have up to 64 simple oscillators. Combine that with the Unison, and you can get a huge sound very easily. You probably won’t ever have to use that much all at once, but it is available.
Wavescanning lets you load up to 16 waveforms, and morph through them in sequence. You could for instance, map the Wave Index control to an LFO, or map it to the Mod Wheel, and control it manually.
Using the Vector oscillator, you can load four separate waveforms and control the mix of them with an X/Y pad if you’d like. You can also individually set the pitch of each one of those waveforms.
In addition to all of the above, there are four modulators, which are actually sub-oscillators. Used in conjunction with the regular ones, you can use these to setup phase, frequency, pulse width, amplitude, and ring modulation. You can load the same waveforms in to these modulators, just like with the regular oscillators.
Filters and Effects
SynthMaster offers ten different digital filter types and nine analog types – no shortage there. The filters vary in sound as there are so many types available. You can get a really smooth Moog type of filter sound, which can self-resonate at high resonance levels. Or use the Dual filter to get some different/interesting results. There is also an optional distortion built-in to the filter section. You can set it up so it’s before, after, or even inside the filter.
There are a good amount of effects to alter your sound. There’s Distortion, Lo-Fi, Ensemble, Phaser, EQ, and even a Vocoder. You also get a Compressor, Chorus, Tremolo, Echo (delay), and Reverb. I thought it was a little confusing the way in which the controls for certain effects are not on the main effects page. To get to some effects, you must click on FX1, FX2, or FX3. This is not a huge deal though, and makes it so the main page of effects is easier to read, as they aren’t all squished together.
There is a great effects routing section with five inserts of effects per layer. Also, a Global Effects section is included, with two busses and up to five effects on each bus.
I found this a very useful addition to SynthMaster. As an added bonus, you can also control effects using the Mod Matrix. Map the Mix amount of the Reverb to the Mod wheel for instance, or the Drive amount in the Distortion section to an LFO, or nearly anything else you want.
Modulation Matrix and Easy Controls
The Modulation Matrix is pretty easy to use. Let’s say you want the filter cutoff to be modulated by an LFO. Easy! Right-click on the cutoff knob, and in “Modulation1 Source”, you pick what you’d like from the selection menu. Then it will appear in the Mod Matrix menu on the right, assigned the correct way. To get it to actually affect the sound, you just turn the bipolar knob.
You could also right-click and assign cutoff (or some other parameter) to an Easy Control. These let you have access to your most used controls, all in one easy to get to spot. You can also label them how you’d like. To get to the Easy Controls, click the Browse or Presets button at the top.
Right-clicking a knob is also how you get to the MIDI learn, and then assign it to a controller.
Arpeggiator, LFOs, and Envelopes
SynthMaster has its own very capable arpeggiator, with up to 32 steps and many modes available. Each step in the arp can have its own note number, length, velocity, hold parameters, and slide. There is also a way to import a midi sequence into the arp.
There are eight envelopes for you to use per layer, including four of the regular ADSR type. In addition, there are two 2-D and two Multistage envelopes, each with up to 16 points each. Four Keyscalers per layer are also at your disposal. To put it lightly, that is a lot of control for sculpting the sound.
With the Multistage Envelopes, you get the same sort of functions as a regular ADSR, but you can have up to 16 points. There is also a built-in looping function in both the 2D and Multistage envelope types.
There are 2 LFOs for each of the 2 layers, plus 4 other Global LFOs. The LFOs can use the regular waveform shapes (Sine, Square, Triangle and Saw), or you can set them to “Step” or “Glide” mode, and you have up to 32 steps for each one of those. You can sync them to your DAW’s tempo, and they also have controls for the phase and speed.
Preset Browser and Partial Presets
There is really just so much in SynthMaster to go over in one review. I like that though, as I’d much rather have tons of options available to me that not enough.
I do have to mention a couple other features: the Preset Browser and the Partial Presets. The Preset Browser lets you look up presets by Author, Style, Attributes and Instrument Type. If you click on Lead, Dance, and Rob Lee for example, it will bring up…you guessed it; all the lead/dance-style presets by Rob Lee. If your DAW is connected to the internet, you can also upload/download presets to an online database of presets. The only thing I think that might be a good addition for the browser is some sort of rating system for the presets.
I really like the Partial Presets option. It offers the ability to save settings for each area of of SynthMaster you’re in. For instance, say you setup a really nice Additive osc with all 8 oscillators set a certain way (pan, detune, etc.) and want to save just that part of your preset. Just click Save, and then you can name your partial preset. Then you can easily load it back up at a later date into a new preset you may have already started. A great time-saver. You can also copy and paste settings from one oscillator to the next, or even between the two layers.
I already mentioned a couple things I’d like to see added or changed, but they are trivial in comparison to the sheer amount of what’s already there. The price for SynthMaster 2.6 is easily a steal, and this is one synth that is too good to pass up.
The quality of its sound is excellent. You can use different quality settings in SynthMaster, each has a different internal sampling rate: The available choices are Draft (*1 sample rate), Good (*2 sample rate), Better (*3 sample rate) and Best (*4 sample rate). The CPU usage is pretty good for most of the standard kinds of presets. Just like most synth plugins; if you use a lot of modulation and have unison going, it can be a little high. I loaded up seven instances and had Battery 3 running as well with no problem, and that’s on an old dual-core PC. Even if the CPU usage is a bit high on a preset, you could always turn the quality setting down temporarily while working on a song. Then you can just crank it back up when rendering out your music.
SynthMaster is also skinable, although I haven’t tried the included editor. There are some optional skins to choose from.
As I mentioned earlier, there are many optional preset banks available for purchase on the KV331Audio website. You can also get some (or all) of those preset banks bundled with the synth as well.
Download the demo yourself, and give it a try with your own system. Also, listen to the great audio demos that are on their site.
What else can I say about SynthMaster? This one should definitely be on your synth shopping list, if you don’t already have it, that is. With it’s awesome feature set, and great price point, it’s a definite no-brainer.
Warren Burt has been involved with pushing the boundaries of musical tuning, timbre, and technology since the early 1970s. We talk to him to discover what motivates these pursuits.
by David Baer, July 2013
Warren Burt has been involved with pushing the boundaries of musical tuning, timbre, and technology since the early 1970s, as well as with designing and building new acoustic instruments for both community and professional performance. Coming from the non-commercial experimental art-music tradition of the early 20th century, his work represents a different point of view from the technology-based experimental dance-music scene.
He is both a composer and performer of music and multimedia. Born in the US in 1949, he moved to Australia in 1975. He has written music for orchestras, choirs, chamber groups, community music groups, as well as for every generation of music technology from the 1960s to the present. In fact, his work with Australian composer Percy Grainger’s “Free Music” had him reconstructing some of the earliest (1948-52) electronic music technology and making live music with it.
In Australia, he worked at La Trobe University, Melbourne from 1975-1981, as well as being one of the founders of the Clifton Hill Community Music Centre (1976-1983), a non-academic, non-pub-based, non-pop-music community music making facility, where an entire generation of Australian experimental musicians cut their teeth in performance and composition in a safe environment away from either the restrictions of academia or the commercial demands of the pop scene.
He was a freelance composer from 1981-2000, working in many areas of the world, doing musical installations, electronic music performances, music for radio (radio art), and an ongoing series of collaborative performances with dancers, actors and poets. He taught at the University of Illinois in 2001 & 2002, finished a PhD in microtonality (new forms of musical tuning) at the University of Wollongong in 2007, and currently composes and teaches in Melbourne, at Box Hill Institute and at Bendigo TAFE (technical and further education college), as well as pursuing an active performing and composing career.
Recent major works include 3 CDs of computer music Etudes, available for free download from his www.warrenburt.com website, music for choreographer Tess de Quincey’s “Moths and Mathematics”, a large work for dance, music and live computer graphics; and “Experience of Marfa” an composition for eight-channel electronics, and 40 voices and percussion, in collaboration with composer/painter Catherine Schieve, and the Astra Choir and Astra Improvising Choir, led by John McCaughey and Joan Pollock.
Most of his work involves working live, in real-time, with technology of one sort or another. He is very involved with getting electronic music out of the studio, out of the bedroom, and into the larger world of music making and interaction with people.
Sound Bytes What kind of music would you be pursuing if this was, say, the year 1913 … no computers, and no notable electronics?
Warren Burt What would I do if I was in 1913?
– Look up Erik Satie and tell him to stop drinking so much – he needs to live beyond 1925.
– Go to Charles Tomlinson Griffes and slip him some antibiotics from the future so he doesn’t die at the age of 35 in 1920 from flu complications from the previous year. I want to see how his transition from impressionism to atonality would progress.
– Go to Charles Ives and give him some blood pressure medicine.
– Look up Percy Grainger and discuss his ideas of “free music” with him. Maybe we could have gotten the synthesizer 40 years earlier, if Grainger had been encouraged.
S.B. Interesting responses, but I was asking what you’d personally be doing musically without today’s technology?
W.B. Well, the reason I gave personal responses about individual people is that I relate to those composers as if they were family. My grandfather, who I was very close to, was born in 1891 (he died in 1968), so I did have a living connection with that era. I relate to those guys as I would to people of my grandfather’s generation, or a little bit older. But in answer to the musical question, probably I’d be convincing people to invent musical instruments that can play microtonal scales. Why should we have to wait 23 years for Harry Partch to do that in 1935? But I then have a further wrinkle on your question – are we talking about me having stepped into a TARDIS and travelled back to that era, or are we talking about me having been born, let’s say, in 1891 and that I was raised in that era and was steeped in its culture. Let’s answer both questions:
If I were a time traveler to back then with all my knowledge intact, and it’s 1913-1916, I’m heading to Paris to study composition with Ravel and learn how he wrote the “3 Poems of Stephan Mallarme”, then I’m heading to Vienna to study with Schoenberg to learn about “Pierrot Lunaire,” then I’m heading to New York to learn about Griffes’ “The Kairn of Koridwen,” and maybe I’ll hang out with Henry Cowell in California and work on my tone-cluster technique with him. So if I were a time traveler I’d like to work with those people who were making very complex, very sensual music at that time.
On the other hand, let’s say that I was born in Coshocton, Ohio in 1891 (maybe in this imaginary history I’m my grandfather’s imaginary twin brother). Now we’re talking about history, economics, opportunity, etc. Would I even have gotten into music? But assuming I did, I’d be thoroughly immersed in the culture of the day. If my penchant for the unusual survives a liberal Midwestern upbringing of the early 20th century, then maybe I’d end up in Minnesota working with John Becker or Carl Ruggles, two early-20th century modernists. Or maybe I’d just end up as a salon-music playing hack. In any case I’d be hoping that my attraction to the unusual in this era would translate into an attraction to the unusual in that era, and I’d be seeking out the most advanced musical thinkers of the day, as I’ve tried to do in this day. (Of course, if I were born in 1891, I’d be prime draft-bait for the First World War, so avoiding that would be a tricky one, too, but I guess we’re just dealing with music and not all the messy realities of living in that era.)
Maybe my music would have evolved into a combination of Ruggles-like atonal melodies with Satie’s irreverent wit and humor, combined with Ives’s juxtaposition of several musical worlds at once.
On the other hand, Gordon Mumma, the electronic music pioneer, is the nephew of Archie Mumma, a salon-music composer from the turn of the 20th century, who was interested in music made from bird-calls. So you never know how that biologically transmitted talent stuff is going to work out. In fact, if I had been born in Ohio in 1891, maybe I would have ended up working with Ohio-an Archie Mumma! (http://www.spencerserolls.com/zencart/index.php?main_page=product_info&products_id=400294400)
Tangentially, I sometimes ask audio engineering students the question – would rock and roll have evolved without electricity? I recently read about compressed air being used to make Edison cylinder and early non-electric phonograph playing very loud (120db) without much control, so perhaps we would have had a loud popular-folk based music evolve without the need for current in wires. This is all in the line of alternative histories, and if they would be possible at all.
S.B. What, no AC/DC? Oh well, no big loss really.
Clearly your musical interests focus on what most people would label “non-mainstream” or “cutting edge”. Were you always “there” or did you gravitate to such things after a more traditional musical education?
W.B. Well, regarding AC/DC, as I said, there were people who were using compressed air to make extremely loud music using acoustic phonograph technology, so AC/DC could easily have happened. Just imagine a stage with 20 or 30 steam powered compressed air loudspeakers, all going at 120dB. That would have been pretty amazing.
I wouldn’t say that what I was doing was “non-mainstream.” I unapologetically feel that I’m a part of the art-music tradition as different from various “pop-music” traditions. But over the course of the 20th century, the balance between the traditions has shifted dramatically. That old spy-master and musicologist Henry Pleasants recognized this as early as the 1940s with his book “Serious Music and All That Jazz.” Today, I don’t think there IS a mainstream any more – just lots and lots of different musics, some of which make money and some of which don’t. But if we’re going to let economics determine our aesthetics (like a lot of the society, alas, does) then we might as well just give up now. Which I refuse to do.
The question you ask though relates to an assumption many people make, which is that in adolescence, you identify with a certain kind of music, and then you are conditioned to think of that as “normal” and everything else as “weird.” And then you struggle to expand your world view to encompass more than the music you liked in high school or something like that. That never happened with me. I played accordion as a young kid, and with the accordion (in the 50s) you played everything – classical, pop, folk, and they all seemed equally valid. Also, my parents didn’t like rock-n-roll, and I didn’t get a lot of exposure to it until I was in my last year in high school. And further, I went to a military academy for high school, so I never really “got socialized” and made a particular kind of music part of my identity. In high school I was already engaged in what I called at the time my “search for the weird.” When I started college in 1967, at the State University of New York at Albany, which was beginning to have a thriving new music scene, (and new poetry and multimedia and video art and electronic-interactive dance scenes as well), I found my “weird.” And I felt so much better in an intellectual environment like a university. So if I ever got “socialized” it was in university, when I embraced being part of the art-world and it’s most exploratory wing. And it was in university that I first was able to play with a very large Moog synthesizer (the CEMS system designed by Joel Chadabe), as well as observe friends do early experiments with using a computer to write texts and make drawings, etc. So right from the beginnings of my artistic career, I was involved with things on the bleeding edge of technology.
And not just mechanical technology either, but also with aesthetic ground breaking as well. As an undergraduate, I managed to meet and talk with people like John Cage, Karlheinz Stockhausen, Gyorgy Ligeti, Robert Ashley, Kenneth Gaburo, Lejaren Hiller, Salvatore Martirano, while studying with Joel Chadabe, and this was the period when many of them were doing their most experimental work. So that kind of thinking didn’t seem strange or challenging to me – it just was the way people were thinking.
S.B. OK, two part question. Since you do teach at a university, I assume you acquired some academic credentials along the way to make you qualified for that position. Do your academic qualifications align with your musically “exploratory” proclivities, and to what extent are you permitted to draw upon those experiences in the classroom? Also, do you find the academic environment especially conducive to your extracurricular musical pursuits?
W.B. I did my BA at the State University of New York at Albany between 1967 and 1971. While there I studied composition and electronic music with Joel Chadabe. I had access to their very large Moog synthesizer, and worked with a lot of guest visitors, as outlined above. I then did my MA at the University of California, San Diego between 1971 and 1975. I worked mainly with Kenneth Gaburo and Robert Erickson there, but also with a number of other people on the faculty. Off-campus work with cellist and Yiddish music scholar Ronald Al Robboy and composer/environmental artist David Dunn was also crucial for my development. So my entire training was in making experimental music and working with technology. In my teaching now at Box Hill Institute (not a University, but a Technical College, to keep things accurate), most of my training is directly applicable to my teaching, although I do teach a couple of straight history courses as well (I’ve even managed to make learning about the 19th century fun!). My composing is a direct outgrowth of my training, but has extended the ideas that were prevalent in the early 1970s. If people want to see some of my work, they should go to my website, www.warrenburt.com, where I have links to my YouTube channel, as well as reports of music I’m making, and some sidebar features about my early history with music technology, etc.
S.B. Critics of the type of music you champion have been known to dismiss it as emotionless and/or purely intellectual. How do you respond to this kind of criticism?
W.B. I usually respond to those kinds of questions by pulling the questions apart. For example, if the criticism is about something being “purely intellectual,” I ask the person to define “intellectual,” “emotional,” what they think is the difference between the two, how does this affect their listening, how DO they listen, what triggers things off in them, etc. A lot of time, I find that people actually examining HOW they FEEL makes them very uncomfortable. Easy, familiar emotional responses are usually brought about by familiar material. A lot of people just want to feel comfortable with what they know. I make it clear to people, in a very friendly way, that I’m more interested in exploration of new material and new emotional reactions, and that the journey is a friendly one, and that they’re more than welcome to come along for the ride. As I sometimes say to my students, “There’s music you know how to write, and there’s music you don’t know how to write. I enjoy making music I don’t know how to write because it seems to be more fun and more of a challenge to do that.”
Then, if they begin to get interested in a more positive way, I can tell them about some of the ideas behind my work – such as algorithmic or generative music, in which I use various systems (like DNA protein patterns) to make music in order to hear what those patterns will sound like, or the ideas of working with new tuning systems – and explaining why one would want to work with tuning, anyway, or working in an interactive way with intelligent machine composing systems – that is, inventing different kinds of improvisational partners than my carbon-based biological friends can do, etc. Once we get talking about that kind of stuff, their original criticisms seem to fall away, and they begin to understand about the idea of music as a way of exploration, as well as a way of familiarity.
Three examples of my recent work with live performance of computer things can be seen on my website at http://www.warrenburt.com/journal/2013/5/13/concert-at-box-hill-institute-may-9-2013.html. In these pieces, I’m dealing with sound poetry, microtonal scales, movement controlled music machines, and interactivity with both a phenomenally talented human (Craig Schneider), and a pretty talented machine (my netbook with AudioMulch, ArtWonk and Glitch2).
Eduardo Tarilonte is a well known sample library developer, and some might argue he is absolutely one of the best. He shares the secrets of what lies behind his specialty.
Soundbytes Magazine is starting new series of Rookie articles, where well known developers will unveil their secrets and voodoo tricks explaining to us, mortals what lies behind their specialty trying to explain the steps that we need to do if we want to follow them, giving us an advice or two, or just explaining what the hell is going on under the hood of their niche. We are all playing some instruments, or use some effects, libraries or loops. We constantly record something, doing mastering or mixing our songs, but do we really know how all those tools that we use is really made? Which skills do we need? It is a time to know better each other.
Eduardo Tarilonte is the first one who will take us in a secret world of sampling. He is a well known sample library developer, if you ask me, he is absolutely the best one. There is no way to spot the difference between the real player and the passage recorded with instrument from one of his library. He started his carrier with Bela D Media, currently working for Bestservice, which is number one Sample library provider in Europe and also highly presented in America. Eduardo got all sort of awards. Sample library of the year, Sample developer of the year, the most innovative sample developer of the year. Just name it and he have it.
More about him and his libraries you can find on http://www.samplelibraries.com/
Sampling for Rookies – How to …
When Arsov asked me to write an article about making sample libraries for newbies, I initially thought to myself…well, it is easy to explain, it is like if you ask a painter, “how do you paint”? If I were a painter, my answer would be “take your pencil and start…”, that’s all…but then, there wouldn’t be any article to read.
But it is not easy to explain, at least for me, an endeavor with so many nuances during the process. On one hand, you can sum it up in one sentence: “start building YOUR dream”. A skilled painter paints as if it were an easy task. But if you ask him why he uses one or another color, he will probably say: “I use this one because it is the right one!”. In other words, it’s easy to say, but it’s still a challenging thing to explain.
Developing a sample library is a primarily creative process, although many people may think it is mainly technical. Of course it is technical, but that is just the part that can be “easily” achieved. The creative part is, without any doubt, the most important one and the one which makes the difference. It must be there from the very beginning to the end.
As I always say: all things that go beyond technique, in other words, those things that are intangible, are the most important ingredients.
So let’s start!
1. Think of something you love, something you would love to sample, no matter if it already exists, once you do it, will be different from others, it will have your seal.
2. Don’t copy the way others develop their products! It’s not because it is forbidden. It’s because that’s very easy to do and will be a waste of your time. Making things your own way is the trick to making a great product. No matter if some people say it is a bad idea, it won’t sell, it will waste huge amounts of time, the idea is too niche, there are other amazing libraries featuring those instruments … whatever. Developing a sample library is actually a lonely process, so get ready to fight against yourself. If you survive, it will increase your self-confidence
3. Once your idea is crystal clear (no way to go back or run away), your project must start with one of the most important quests...finding the right musician. This might seem an obvious statement, but it isn’t. You are capturing the soul of the musician in your sample library.
Avoid amateur musicians, unless they are very talented, but on the other hand, be careful of musicians that are too professional, or they will play what they want, not what you want!!
Your role with the player is like a film director with an actor. Listen to him and his suggestions first. If he is good enough with his instrument, you may learn some details that will be useful to sample. But later, ask him to play what you finally want. It’s one thing to play the real instrument, but quite another is playing samples on a keyboard later on. This is not something easy to explain (again), and of course depends on the instrument being sampled. For example, the player may prefer playing a hard attack in every sustained note, but such an attack might sound harsh in every note when you play it in your keyboard. What you should be going for is even transitions. You can record them in many different ways; everything will sound “real” when you listen to the original sound played by the musician. But once you bring them to the sampler, things change. That’s why selecting exactly what nuance you want to sample will make the difference. You are the director, never forget it.
4. The recording sessions…well, this is probably the most annoying part, listening to zillions of notes, one after another. But you must pay attention to every single note to be able to go home with good samples to edit. Try to use the best gear you can. Choosing a great mic is important for the final sound. Good gear is a must, otherwise the sound you will get won’t be as good as everyone expects.
5. Now you have TONS of audio data on a hard drive (you better have it on a couple of drives at least … hard drives fail). And now starts the “fun” part … EDIT!
You can do it in several ways: automated processing, hiring or asking someone to help you edit and name all the files, or the only way in my opinion: doing everything on your own. Otherwise you won’t really know if the files others have selected are the right ones. If there is something weird on them, that will be noticed for good or for bad in the final result, which is what matters.
It is important that all samples are coherent from the first one to the last one (this step has also to be taken into account during the recording session). Otherwise, when you finish your puzzle and put all pieces together, might sound weird, with some similar notes having a different timbre or character. But if you do it right, your samples will bring to life a real instrument.
As editing samples requires a long daily schedule and is a mechanical job, be careful to have some rest from time to time. Otherwise you will damage your back, shoulders and wrists. Unfortunately, I can tell you about that.
6. After that comes mapping and programming. Programming it is important, as it will allow you to make the instruments playable in the way you want. But don’t rely too much on scripting or programming, or your will get a robot instead a “real” player in the end.
7. Ask for some beta testing and feedback from sincere friends. But don’t despair if you don’t hear what you expected, you will know if they are right or not. In the end is your general vision which counts, even if the world is against you!
8. Use it in a real composition. Make some demos to test how far the library can reach. That’s the real test. That will help you to fine tune the patches. Fine tuning is important and a more gratifying task, like mixing after composing.
9. Spread the word! Let the world know that your library exists. No matter how good it is if nobody but your friends knows about it.
10. And the most important … just before wrapping up your sample library, start developing another one!!! Developing sample libraries is absolutely addictive! So handle with care.
And to finish…don’t think too much about the profit you will earn selling your library, think about how good it is gonna be during the whole creative process … and it will sell!!
This is just my point of view; others will have different visions … create your own.
Come on…stop reading and start developing something! 😉
Before starting recordings, it is important to choose the right mic placement. Spend some time testing which position gives the best sound for the instrument.
Make sure you take some pictures of the player, mic placement, gear settings, etc. for future recording sessions with same player and instrument (fixing mistakes, recording more samples).
Be careful of the background noise, especially for plucked instruments, where the release must be maintained for seconds. The noise in samples is multiplied by every note you play together.
Ask the player to come to the recording session with comfortable clothes and avoid watches, rings or any other items that can make undesirable noise during the recordings.
You must have a clear idea of the general vision of the library before starting. That will guide you to know what exactly you need to sample and what not.
In this tutorial we to look at two important mixing factors: computer sound internal formats and sound levels and explore how they’re related.
by David Baer, July 2013
I pulled one of my CDs out the other day that was released in 1998. I noticed that the cover proudly announced “recorded in 20 bits!” I had to chuckle since these days recording in anything less than 24 bits is regarded as unacceptable. But it also made me recall how back then, at a time before I had done any studying of digital sound technology, I was completely puzzled by what that meant. I knew CDs used 16-bit encoding, so just how the heck did they stuff those 20 bits onto the CD? What was the point?
Now, of course, it’s easy to grasp, but that’s only after I’ve spent many hours on computer sound forums and read numerous in-depth explanations of various aspects of digital recording. And the time on the forums has made it clear that this topic is one most digital sound enthusiasts struggle with at some point in their education.
So, in this tutorial we’re going to look at two things: computer sound internal formats and sound levels and explore how they’re related. Those of you who are already familiar with computer fixed and floating point data representations will probably want to skip the first part of the discussion, but still may be interested in the second where we’ll look at the relationships between the formats and the implications they have on sound levels and recording/mixing practice.
What we’re not going to get into is recording rates or compression issues. We’ll keep things simple and just assume everything is recorded 44.1K samples per second, just like the data on a CD. Just accept that that rate is sufficient to encode sound that encompasses the range of frequencies audible to humans. Also assume we’re interested only in recorded sound that will be released in CD format. Consideration of compression schemes like mp3 would needlessly complicate the discussion at this point.
Fixed Point Representations
As mentioned above, CDs use a 16-bit number to represent one sample of sound for one channel. Before we look at wave files, where there are more options, let’s understand exactly what that means. 16 bits can represent exactly 65,536 discreet numeric values. We can represent integers between zero and 65,535 for unsigned numbers and between -32,768 to 32,767 for signed. Or, and this is the way things work with digital audio, they can represent 65,536 unique output voltages coming out of a digital to audio converter (DAC) in your sound card.
So, say your sound card outputs between exactly -1.0 and +1.0 volts for a total range of 2.0 volts, the extremes will correspond to interal values of -32,768 and 32,767. Each increment of 1 in the 16-bit value represents an increment of 2/65,536 volts in the output. That is to say, we have 2 volts divided into values that are 2/65,546 (or 1/32,768 if you prefer) volts apart in magnitude.
But, you say: there’s no number that corresponds to exactly zero volts using this scheme. Aha, well spotted. While true, in practice this is unimportant. You’ll see later that at 90dB down from the maximum, not having an exact zero value is going to be utterly insignificant.
If in the process of recording sound we capture 16-bit values, then we want to use most of the available bits to capture as much detail as possible. The release format is going to be 16 bits one way or the other. If we don’t capture a full signal, detail will be lost. A mathematically rigorous explanation could be offered, but really this is just common sense.
The fixed point numeric format has non-negotiable maximum positive and negative magnitudes. If we try to assign a number that exceeds those limits, high order information is discarded leaving unpredictable values behind. When this happens, it’s called clipping. It can sound something radio static being sent over a dodgy connection to a speaker with a damaged cone. So, long story short, 16-bit recording/mixing should be done using Goldilocks levels: just right – high enough to capture detail but not so high as to clip.
Fortunately, these days we can use a 24-bit fixed format for representing sound data … up until the time it gets distributed on CD where it necessarily needs to be trimmed to 16-bit. Happily, sound cards with 24-bit analog/digital conversion capabilities are well within the price range of even amateur enthusiasts.
With 24-bits, our value range is between -8,388,608 and 8,388,607. This does not mean that we have a signal that is 256 times as strong as with 16-bits. Our hypothetical sound card still produces between -1.0 and +1.0 volts for the two extremes. What we have is 256 times as much detail. The upshot of using 24 bits for recording is that we can stay comfortably below levels where clipping occurs and yet still have more detail in the signal than can be retained when we finally convert back to 16-bit for release.
The wave file format is flexible. It supports 16-bit value scheme identical to CD data values, but can support other formats, including 24-bit and floating point which we’ll look at next. Other sound file types also are flexible in this fashion. The point is that we can retain 24-bit recorded sound in files as long as we want on our computers. It’s not until we want to do CD distribution that reduction in number of bits will enter the picture.
Computers have an alternative numeric representation that is preferable to fixed point for many types of calculations; it is called floating point. Consider the number 1,234. We can also write this as 1.234×1000 or 1.234×103. This is the idea behind floating point. There’s a base number that’s always close to 1 and a multiplication factor, called a mantissa. Floating point comes in two varieties: single and double precision, which use 32 and 64 bits respectively. The term “floating point” without a “single” or “double” qualifier normally implies single precision. The range of magnitudes representable using even single precision floating point is between incomprehensibly small and incomprehensibly large. The double precision limits are even more incomprehensible.
24-bit fixed numbers can be thought of as having 23 numeric bits plus a sign bit. For comparison, floating point offers 24 numeric bits and double precision offers 53. So, floating point offers more detail than 24-bit fixed. But wait … there’s more. There are always 24 bits available for recording detail using floating point. If you’re dealing with low levels, you are not “wasting” high order bits as you would with fixed point. As to high levels, with floating point clipping is essentially nonexistent!
Clipping occurs using fixed point when the values needing to be represented exceed the capacity of the format. But with floating point, numbers can get arbitrarily large, so as long as you remain in floating point, clipping is effectively impossible. Now, the moment we convert back to fixed, making sure the conversion does not produce clipping is imperative. But at least we only need to worry about it at that one point.
So, does this mean you can run floating point signals through your DAW and not worry about gain settings until the final output? To a certain extent this is true. But your metering will be difficult to read (although we may not care about metering anyway if we’re ignoring levels). A more compelling reason is your control over a mix. Sit down at your computer and bring up your DAW (or just look at the pictured SONAR faders). Set a fader at zero and adjust it either way by 3dB. No problem. Now set the level down to, say, -30dB. Try the 3dB adjustment now. Not so easy, I think you’ll agree. So, even with the safety net of floating point, a Goldilocks approach to things has advantages.
Formats and Levels
At this point you have a sufficient understanding of the capacities of the available numeric data types. Tying this knowledge to the concept of decibels should fill in the remaining piece of the puzzle. The decibel (more commonly written “dB”) permeates the language of sound technology. By itself, the dB does not connote a unit the way “gallon”, “meter” or “ounce” does. It always denotes a relationship between two values of like units or, if you prefer, a ratio. Those units can be volts or watts, air pressure units or anything else.
The dB is a logarithmic unit. Don’t know what that means? Don’t worry. It works like this:
- “B is 20dB less than A” means B is 1/10th the intensity of A. Alternately we can say: “B is 20dB down from A.”
- “B is down 40dB from A” means B is 1/100th the intensity of A.
- “B is down 60dB from A” means B is 1/1000th the intensity of A.
- And so on …
Note that the above numbers are valid for voltage and for sound levels of tracks within a DAW. Power (watts) is another story, but the explanation will have to wait for another time.
In your DAW and other sound software, scales will often be calibrated in dB, such as the faders pictured earlier. You will probably not even see “dB” anywhere, but it’s usually obvious because the numbers are not evenly distributed on the scale. Here the level 0 is associated with the maximum signal before clipping happens (or would happen if the underlying numeric format were fixed point).
Again, dB implies a comparison of two levels and it’s always important to understand to what the comparison is being made. Most places in sound software, the implication is that zero on a dB scale means “loudest advisable”.
To illustrate, consider the figure of the single cycle of a saw wave in SoundForge. The scale on the left is clearly logarithmic and the maximum value is 0 dB. Note also we have 0 dB pictured at both the positive and negative extremes. I saved this single cycle as a wave file (one channel, 16-bit), extracted the data into a spreadsheet in Excel and graphed it. The slope is just about what we see in the SoundForge depiction, but in this case the y-axis units are the fixed-point values encoded in the wave file. So, you can readily see that whether we’re using dB or integer values, the essential information is the same.
The Most Important Things …
Let’s take a brief detour and consider the compelling demonstration that can be seen/heard here:
This article by audio authority Ethan Winer demonstrates, with audio examples, how a signal that’s down 45 to 55 dB in a mix becomes inaudible. This is true even when the lower-level signal is blatantly irritating and the higher-level content is smooth music devoid of percussive transients. This is such a useful piece of information that you should never forget it when making day-to-day decisions in your tracking and mixing.
Let’s look at some further basic truths. The following figures were derived using the dB Voltage Ratio calculator available at:
Our DAW quantities are digital numbers, not volts, but they end up in volts coming out of your sound card, so a voltage calculator (as opposed to wattage) is the correct one to use.
The largest value that can be encoded on a CD is over 90dB greater than the smallest in magnitude (closest to zero). Thus, we see how the commonly quoted fact that the dynamic range of a CD is 90dB was arrived at.
Likewise, we can see that a 24-bit format (23 bits numeric plus sign) offers an immense dynamic range of over 138dB, well in excess of the range of human hearing in the most capable of listeners.
Using 20 bit fixed format vs. 24 bit causes us to lose about 24dB in detail. But we’ve still got a dynamic range of 114dB, comfortably in excess of what can end up on a CD.
So, finally we are armed with some iron-clad facts we can apply to making informed decisions about recording practice.
Let’s assume we want to record to 24-bit fixed point. If we set our peaks at 24dB down from maximum, we’re effectively keeping the high order four bits for headroom, and as a result, we’re accepting that we have four fewer bits at the other end for detail. But even with 20 bits of usable precision, we’ve still got 4 bits of “extra detail” compared to 16 bit format. In practice, however, it’s far more common to leave about 12dB (or 2 high order bits) of headroom, which is normally plenty for unanticipated peaks.
Your DAW may give you the choice of using single or double point floating point for internal signal processing (even if the tracks are stored externally as fixed integer). While state-of-the-art PCs can handle double precision without too much extra strain, it does require more processing power, memory, and if exported as such, more disk space when using double precision. How much does using double precision buy you?
Figure it this way, in single precision, you’ve always got 24 bits, irrespective of signal level. Eventually that will be reduced to 16 fixed point bits for final export. Computations in effects processor calculations will result in some level of round-off error. Staying with double effectively reduces that accumulated round-off error to zero in the final export. But you’ve already got a “buffer” of 8 bits in single point floating with which to shield the significant high-level bits from seeing the effects of the round-off error. So, I’m not asserting that you will never, ever, ever hear no difference between using single and double floating point. But I am suggesting that the majority of time, common sense tells us there will be no audible difference.
This is not to say that you’re not completely avoiding double precision when you tell your DAW to use single. Some plug-in developers will determine that double is advantageous or even essential in calculations for their purposes. In such cases they may use it “under the covers” without your knowledge or consent, converting back to single precision only when passing the results to output ports.
Finally, there’s the subject of dithering. This is a technique wherein very low-level noise is added when reducing the number of bits, usually when exporting from our DAW. The low level noise is in the form of random assignment of values to the lowest level bit in each data value. Some people assert that dithering, while it does no harm, is completely unnecessary. Others claim it is important and a few individuals have even boasted they can tell whether it’s being used or not. Well, let’s see. Dithering would be 90dB down from the loudest signal on a CD, and let’s say 65dB to 70dB down from a typical level. But Ethan Winer has shown how even at 45dB to 55dB down, a weak signal becomes inaudible in the presence of a stronger one. So, go ahead and agonize over which dithering algorithm you’re going to go with if you’re truly concerned. Personally, I’m not going to be losing any sleep over it, one way or the other.