Sound Vacation part 2

When most people take a vacation in the truest sense, it means much more than a simple absence from the job. The word "vacation" stems from "vacate," which means in Beckham speak "Get Out!"

Considering that I've been working only in Thor (and FM8 as a analysis tool), it's only fitting that I completely get out of sticking with that synthesizer and move on to something else. I knew NN-XT was a joy to use before. Spending all my time there reconstructing the Casio MT-205 is something like a sound safari. I'm thinking about all the techniques I simply ignored or laughed at and taking them much more seriously. So far I'm pleased with the results.

The great thing about getting out and exploring sound in new ways is the time spent coming up with new ideas for future work. For example, the work I'm doing on the Casiotone involves distilling the decay and sustain phases down to a single waveform. Why not take that a step further? Using Absynth, I could analyze any given sample and split it into 12 different waveforms. With Absynth's morph waves and double oscillators, I could easily make wavetable versions of these same instruments! I doubt I'd use all 12 possible waveforms--the job should be done with 4 waves at most. Another possibility is using Absynth's sample-playback to get just the attack portion of the sound and fill in the rest with re-synthesized waveforms.

There are other lovely possibilities one can take from this. Re-synthesized sounds can serve as templates for other sounds using waves from other sources. Most of my work on the Thor FM sounds will be applied towards new sounds using the same architecture. Perhaps I'll also try transferring those same Thor sounds to Absynth.

Yet another idea I've had related to sampling is to take 1- or 2-cycle waveforms created in Absynth (additive synthesis, most likely) and load those in a sampler such as EXS24, NN-XT, or the Akai. This is similar to what I'm doing with the Casiotone except my goal with Casiotone is to try to capture some of its native chorusing and vibrato. The possibilities and opportunities are great.

So while I am trying to clear my head somewhat, I'm also trying to keep it from exploding sometimes. The MT-205 side-project is going well so far. Perhaps next time I'll report a little more specifically on what I'm doing with it.

Sound Vacation

While I'm taking a vacation from Thor patches, that doesn't mean that I've stopped working. It also doesn't mean I'm taking a vacation from working out ideas and charting possible new courses.

Some concerns that I have relating to my current setup are that I've been too unprepared to work with real synthesizers, which is where this whole thing is going. I should have my Akai S2000 (thank you, eBay!!!) by the end of next week. Yes, I'm perfectly well aware of the limitations of a 32MB system. But that has made me all too aware of my own limitations as well. You see, back in the day I used Gigastudio v. 2.5. I'd wanted to use it live, but with all the clicks and pops in that cheap computer, this just wasn't meant to be. I've been absolutely amazed with EXS24 since I started using Logic. I've also been really impressed with the few instruments I have "powered by Kontakt" and have often wondered if I shouldn't invest in Kontakt. I thought perhaps of getting a new PC for the purpose of running Emulator, which I might still do one day. Now I'm totally addicted to NN-XT in Reason. I like the ease of use, the filters and envelopes, and the overall flexibility/versatility of NN-XT. So yes, that's certainly going to make it into my live setup with Mainstage.

What I've realized, though, is how fabulously spoiled I've become in using these software samplers. I think what I need is a new way of thinking. It's only a matter of days before the Akai gets here, so I think now is the time to get used to a whole different approach to sampling.

I think most of us would agree that the ideal when sampling is to sample as much of an instrument as possible. Take a piano, for instance. You're going to want to sample each note for as long as the note sounds. You want this pedal up, pedal down, upper string resonance, inside the piano, out in the concert hall, and each for at least 4 velocity layers, all at the highest sample rate and bit depth available. OK, so we're talking about an instrument that's going to take anywhere from 4 to 8 GB.

That's well and good if you absolutely CANNOT let that Bosendorfer get away! But it's perfectly unacceptable when dealing with a vintage 32 MB sampler.

So what exactly is Kyle Beckham up to now? Here's my goal for the next week: Sample the entire soundset of the Casio Casiotone MT-205. You might ask, "What's the big deal?" It's mostly sentimental, to be honest. When I was a little kid, my aunt had one of those in her piano room. I'm not exactly sure why she got it, but I do recall thinking the sounds were incredible. This was a tiny "toy" keyboard with a sound that could fill up a room! I always wanted that keyboard. Sentimental reasons aside, there is a nostalgia factor I think anyone can appreciate. It's a bit of a relic, preserving quite well the kinds of sounds that were really popular in the 80's.

My aunt had a nasty smoking habit. The last time (and it's been years) I played that thing it was already having some trouble. As both my aunt and uncle worsened in health, it became obvious that even if the thing still worked, I'd never get my hand on it. Thanks to the miracle of eBay, I found one in perfect condition and have played with it off and on over the last few years.

I've tried sampling it before, but it was the typical "make long samples, play the entire decay," and so on approach that's a killer on memory and processor resources. After completing the sound design course, I thought now would be a good time to return to it with a more sensible and relevant approach to sampling.

The point I'd like to make here is my change in thinking and general approach to sampling. Samplers can often be used as synthesizers in their own right. The S2000 is such a sampler with a wide range of envelopes, lfos, filters, and effects. Lets suppose, for instance, I want basic analog sounds. All I really need is a cycle or two of geometric waves and the S2000 can do the rest (lead, pad, classic synth sounds, etc.). Or if I want the kinds of sounds I get from Absynth, I can just use Absynth to create whatever waveform I want and let the S2000 do the rest. From that perspective, I could program all the instruments I would need for playing live from a single floppy disk.

32 MB begins to look like a lot of memory!

I'm trying to practice this. Just today I reconstructed the Casiotone MT-205 Vibraphone using NN-XT. The way I did that was to record both stereo channels separately. The next step was to create two mono vibraphone patches, one to represent its respective channel. I trimmed each sample so that each retained a strong attack transient and a little of the decay/sustain. When I imported these into NN-XT, I set loop start and end points at the very end of the sample and slowly pulled back on the start time until I had at least two cycles. While the ends of the loops may not exactly line up, there is one important advantage to this approach: The loop pop or click is always cyclical. So as you pull the loop start down towards the beginning of the cycle, the apparent pitch lowers and fades as you approach it. While there will always be some noise with this approach, it is masked by the pitch and timbre of the sampled waveform. By pulling this back to as many as 1 or 2 cycles, the quality of sound improves. You have to take care though, because this can create something that sounds like a sub-oscillator. If you like this pulse sound, of course, it can be shaped with filters to get completely different results than what you might have initially gone for. Personally, I choose to avoid this because it seems to defeat the purpose of sampling waveforms to begin with. So far the only instruments I've made this way are a MT-205 piano (for testing only, but will revisit) and the aforementioned vibraphone. I'm very pleased with the results!

The downside is that Casio seems to use its own reverb/chorus/vibrato for its programs, nothing I can manipulate for the keyboard. Making such a short loop destroys this effect, so I have to compensate by using my own chorus and reverb in addition to making the appropriate settings for vibrato or tremolo using NN-XT's LFOs. It's an unfortunate reality, but it beats searching for hours finding loops points that preserve the original effect! There is so much modulation going on in the vibraphone that it's a next to an impossible task. I'm trying very hard to minimize hard disk consumption during this process, so the good news is that the vibraphone samples occupy less than 2MB of memory. Looking ahead to the S2000 arrival, this means that keeping all single programs to 2MB or less will allow for a multi of 16 different instruments. That's a good thing since the S2000 is 16 parts multi-timbrel. And it will occupy close to the maximum 32 MB.

That's very exciting for me not only as a sound designer but also as a composer and electronic musician. I'll be expanding my setup with real synthesizers and samplers in addition to the virtual ones I keep with me; however, for live playing that's not really a portable option. I'm hoping by changing my approach to sampling and getting some practice in that area I'll be best prepared for developing road-ready instruments.

Halfway Point

I've completed my 64th Thor FM patch, which means I've worked through ROMs 1a and 1b. I want to take some time to review and reflect on the work done so far and consider how I'd like to proceed from here.

Number one, I need a break! As I mentioned in a previous blog, I took a few days just to refresh my ears and break the tedium. My usual process involves more listening than looking at algorithms. Why, aside from the obvious? Thor and DX7 do not handle the same. The settings are so radically different between the two that the ear has to be the ultimate judge when a patch sounds "right." The problem is that the DX7 patches are full of variations, sometimes using completely different algorithms to get the same kinds of sounds. So after repeating the same settings and routings across multiple patches, trying to differentiate between similar-sounding patches, and creating variation among patches that are too similar-sounding, the ear is exhausted. I actually find on some days I need to sit around, turn the brain off, and only passively listen. I also found sitting in a quiet room to be refreshing. I sometimes think I can hear the sonic energy of the day slowly dissipating, fading echoes of all the sounds of the day. I'm constantly surrounded by sound, whether I'm listening to a piano student, engaging in conversation, watching a movie, making a new instrument, trying to keep my kids from killing each other, or even sleeping. I actually have trouble sleeping without a box fan on. Yes, I even require noise to fall asleep! The ears never get a break. So I need some time to clear my head before I listen to new sounds.

Number two: I've learned a few lessons and tricks along the way. Big example is Thor can't reproduce the same feedback loops as the DX7. There are a few options, though. I typically like to route an oscillator to itself, preferably with 0 FM amount and modulating its own pitch. This makes the pitch highly unstable, but it can be partially rectified with keyboard scaling. Also, the effect has to be very subtle. Such a low level can't be effective across the entire keyboard. The effect can also be enhanced by allowing just a little of the self-oscillating signal to leak into the mix rather than strictly using it to modulate another oscillator. This happens because in FM, part of the modulator signal is present in the waveform. You can get closer to a true DX7-style FM sound by mixing the signal of a self-modulated oscillator with the rest of the oscillators, even oscillators that are modified by another.

Number three: Oscillators that modulate the pitch or FM frequency of others create pitch instability and upward shift (downward if using negative values, which we never do). For solution, see Number Two.

Number four: Due to issues with Two and Three, oscillators can mod themselves or each other at such high extremes as to create some characteristically colored noise effects. This is useful for drums and other sound effects related to noise, such as the Train patch.

Number five: The DX7 often includes sets of operators that are almost identical except for subtle detuning. This is modeled after analog detuning and can be replicated using the chorus effect in Thor. Seriously, if it's not necessary to detune pairs of oscillators, why bother? Another approach is taking advantage of the pitch instability of oscillators routed to each other or themselves. I love doing this! But the main idea here is it's not always necessary to use all three oscillators to replicate DX7 patches. If all it takes are two FM oscillators to get the sound you want (or even just one), that opens up the patch for more complexity--such as mixing a percussive bell sound and a pad, or even two completely different timbres. Patches such as the tubular bells, any kind of chime patch, chime/bell patches with flute, strings, and synth pad, are just begging to be tweaked. Every chance you get, look for detuned operators that only serve the purpose of chorusing effects. That will allow for a tight, economical approach to patching which will open you up to new possibilities.

Number six: Use the second filter as a 4th sine oscillator. Avoid using this too often, but don't be afraid to use it when it comes up. Technically, of course, it's possible to add filter 1 as an oscillator as well. I prefer to leave this alone because I may want filter 1 for timbre variation. We tend to associate FM timbres with bright, metallic, cold, digital kinds of sounds, but DX7 presets typically (not stereotypically) are very dark, warm sounds. Feedback loops get a nice analog-like sound. So it's not fair to dwell on the percussive bell and electric piano sounds when there are so many different possibilities. In fact, the DX7 was sold on the idea that its sounds were life-like, more so than analog synths. Unfortunately, with some of the odd routings in pure Thor FM patches, this concept is completely lost and we have to rely on filters to regain some of that warmth. So while using filter 1 is possible, I think it's better to leave it alone. I also don't stress overusing filter 2 as a 4th oscillator because sufficient care should be taken that Thor's resources are used wisely. But I also recognize that many DX7 algorithms leave you with little choice. It's a nice option to have if you need it.

Number seven: In relation to my points about feedback loops, using the shaper is also an option. At issue here is the shaper changes the timbre of all voices summed, which can make for nice guitar distortion. So while this works for some sounds, it doesn't work for all, and care should be taken if using the signal from the shaper as a modulation source.

Number eight: Certain algorithms are best reproduced by splitting them apart and using the sum of the halves to express the full timbre of the original DX7 program. For example if you have this algorithm: A+B:C where A and B are modulators and C is the carrier, you can plug the value of A:C into oscillator 1 and B:C in oscillator 2. This is cool because it allows greater control of FM amounts with your envelopes. The result are often, in my opinion, much richer than the original DX7 and could make for some nice evolving lead and pad sounds.

Those are some of my observations on what has been happening so far. I've resisted the temptation to create variations of these patches. There are just so many possibilities! I do look forward to finishing the next two ROMs, at which point I will refine my current library and create variations on the work that's been done. I really want this to be a good thing, so it really will be fast and furious back to work after I take some time off. I really can't wait!

On to other news...

I've been too content with gigging my computer. That's not likely to stop any time soon with Mainstage having become absolutely critical to what I do playing in a band. But I've gotten too addicted to software samplers for live use.

Here are some drawbacks to soft samplers onstage: Along with processor power used in running a soft sampler, the samples themselves take up a lot of memory. This isn't a problem if your MBP has 4GB RAM installed, but I only have 2GB. Mainstage 2 is decidedly fatter in terms of memory usage than the previous version. Something must change. There are two options here: Create low memory samples and treat them like basic waveforms the same way the big workstations do. Or you can buy a hardware sampler and do the same thing, just take all the number crunching off the computer and have Mainstage do all the patch changes in performance. I've already saved a lot of my resources by moving a lot of what I used to do in Mainstage to Reason. I'm also already putting my sounds to use. While I do like Logic's version of the B3, the FM version I made in Thor isn't half bad. I simply added a Leslie sim to the appropriate channel strip in Mainstage and I'm off and running. I'm thinking now towards doing the same thing, only in a hardware environment.

I took the plunge today and bought a cheap Akai S2000 off eBay today. It's a well-known fact I'm broke, so I'm really careful about these kinds of things. It was a steal! I had to jump on it. So here's the challenge: Develop a library of sounds for a machine with only 32MB of RAM. In a previous life, I thought working under these conditions was totally unacceptable. But the truth is that a good programmer ought to be able to get all the information he needs in a couple of seconds and use the sampling hardware as a synthesizer, molding and shaping the sound the same way anyone else would with a conventional synthesizer. There are certain kinds of sounds I need to produce, and it just can't be done without sampling. So it's a very exciting time. I may be taking the week off next week from designing sounds in Thor, but there will be plenty other work to do.

I also want to say that I'm not without a few secret weapons, at least one which should be here in the next few weeks hopefully. One of those is the Casiotone MT-205. It doesn't have a lot of different sounds on it, but it does have the key sounds you need in a performance or studio setting. To me, that makes it a must-have in a sampling environment.

Eventually I hope to get to some acoustic pianos, electric guitars, and so on. I've got this cheap bass guitar that is very unique-sounding. That thing will work its way into my rig at some point. I also have an extensive collection of bass clarinet sounds I think would be useful. And then there's this pipe organ I really want to get my hands on. All in good time!

I'll blog again after I've done some work. I'm really having fun with this, but it's time for a break. I can't wait for some time off so I can get started on the new patches.

LFO As FM Modulator: Pipes 2

I'm blogging on this patch because I found it terribly difficult to pull off.

Pipes 2 is another one of those patches that has 2 stacks of three operators. The trouble with this patch is in the first stack. Op A has a ratio of 19, B is fixed frequency of 1Hz, and C is 2. I started out by trying to modulate the LFO rate with an oscillator and have the LFO modulate the pitch of another oscillator. I experimented with this in several different ways, and the results were too uncontrollable. The thing is you can get away with using up two oscillators for this kind of thing because you can always use a filter as an oscillator in the second stack. But there was no appeasing the gods on this one.

It took some reverse engineering to pull this one off. I started out by just listening to the first stack. I noticed two things: First, there was some modulating of the brightness caused by a high modulation ratio. That translates here into amplitude or FM Amount. Second, there was a very slight variation in pitch. This means that the second operator is itself acting as an LFO (which was obvious from the start, but how?) and is affecting both amplitude and carrier pitch.

I found this discovery to be exciting. All I have to do is use LFO1 to mod Osc1 FM amt and Osc1 Pitch in very subtle ways. The other stack will need two oscillators regardless, but by eliminating an operator between A and C, there's no need for the "4th Oscillator" effect.

With that in mind, it's almost a very simple patch. First set up the oscillators: Osc1 is 19:2, FM amt. 2. Osc2 8:1, FM amt. 0. Osc3 1:2, FM amt. 0. Next set the LFO rate to 2.02 Hz. This isn't a mistake: Originally the OP B fixed frequency is 1 Hz, but what we hear seems to be positive phase only, or a sine wave with only absolute values. Hence the LFO has to be twice the original setting to replicate the effect (like I always say, this isn't an exact science). We also need a MG to scale each effect according to the percussive chiff of real organ pipes (again, not an exact science!). Set MG A 25.1 ms, D 77.2, R 45.2. Another detail is the timing on all DX7 envelopes are slightly different--modeled after the air moving through pipes, I imagine. Let' set one more envelope, the filter envelope, to A 29.8, D 29.7, S -27 dB, R 23.5. The S level accounts for some amount of FM in the original patch, therefore it should be present here. Amp EG is a standard sustain envelope but accounting for pipe acoustics in the attack transient. Something else I find useful is altering the release transient so that it includes the sound of the pipe cutting off and leaving just a little sine wave action after the initial release.

Now we have everything set up. Let's go to the router. LFO1 to Osc1FMAmt, amount 75, scale 99 Mod Env. LFO1 to Osc1Pitch, amount 22, scale 99 MG. FilterEnv to Osc2 FMAmt, amount 59 (no scale). Osc2 to Osc3Pitch, amount 34, scale 82 MG.

For little details, I added a LPF to filter 1 because I thought some of the noisy components of the sound were a bit bright and harsh. Since you're using the filter EG elsewhere, I recommend turning the env amount all the way down. Chorus could also be used tastefully here, but it is optional. I set this the way I like it, disabled it, and then saved the patch so it will be there if I ever need it.

In conclusion, we're essentially "faking" the sound of a fixed-frequency FM op by setting the LFO to modulate the sonic components that we hear change. This is not a true substitute for the real deal, of course. But at the very least it help you miss it a little bit less!

Double Stack: Harp 2

Previously we've seen how easy it is to include a 4th modulator by tricking a filter into thinking it's a sine oscillator. As I'm often quick to say, it's not a perfect, ideal, or exact science; but it does yield results. In this walkthrough, we'll take a look at a common DX7 algorithm that breaks 6-op FM into a pair of 3-op stacks. We'll be rebuilding the Harp 2 patch.

I've been away from blogging for a while for different reasons. One reason is that my MBP is fixed and I've resumed using it. Partly due to that is the distraction (ironically) of having a computer that works so well--maybe too well. Another reason hand-in-hand with the distraction is not being able to keep focused on programming. Some of the tasks are too repetitive and mind-numbing. I think what happens is I get used to hearing these patches. Even with the DX7, many of these patches are simply variations of others more or less complicated. Thor fails at accurate representation of them. My tendency is to rebuild each patch from the beginning rather than recycling similar patches. It's good for practice. But it's frustrating when working through 6 different guitar patches and they all start to sound the same.

The way to combat this problem for me is to stop relying on the visual part of the software, shut off my brain, and listen while making adjustments. That helps me come up with unique-sounding patches that may not necessarily have been what I was going for but still sound great. For example, I came up with a guitar patch while attempting to model a DX7 patch in a series of guitar patches. I decided I liked what I'd come up with even though it didn't sound exactly like the DX7. Mine was rough and edgy compared to the mellow DX7. The next DX7 patch was rough and edgy, however. I started out modeling that sound and then tweaked some of the FM and routing amounts to mellow it out. So those two are opposite their DX7 counterparts. But so what? Now I have some interesting guitar patches.

I realized at that point I had to get away, effectively giving my ears a vacation. I was ahead of schedule at the time. I'd love to have kept going, but it was just too much. And I did continue with other work--I programmed some pipe organ patches for the Korg T2 at church and spent another entire day programming the new version of Mainstage (Rewire compatible so I can go live with my Reason instruments). But as of last night, it's back to work. After this blog, I'll have only 11 patches to go before I get to my halfway point.

So here we go.

In FM8, I'm looking at Harp 2. In comparison with Harp 1, there's much more op detuning here. My first instinct is to use chorus, but I think I have a better way for our purposes. Op A has a feedback loop. The approximate ratios are 3:3:1 for the first stack. I'm thinking ahead to how I'm going to create this in Thor, so my first impression is that I'll have to use two FM pairs for this part. The ratios for the second stack are 3:2:1. This is making me nervous. I'm used to converting filter 2 into a 4th sine oscillator, but this is typically done as a sub oscillator, a fundamental, or an octave. I don't see an easier way. Filter 2 will have to be at the top of the chain. Otherwise, ignoring the detuning issues, I expect this to be a fairly easy patch.

One little note: Unlike my prior modus operandus, I want the detuning aspect of sound design as a central idea of this patch.

In FM8, I can see that Op A serves as the percussive, attack transient for this patch. I've already made up my mind that this stack will need two oscillators due to Op A's feedback loop. So I'm starting with Osc. 1 with FM amount all the way down to 0, carrier 3. The routing for this is standard procedure by now, though this one will have a little more kick. The router allows for two scaling options, so I'm taking it. In the 3rd router section, set Osc1 as source, Osc1 pitch, Scale 1 Key Note (full range), Scale 2 Mod Envelope. This is necessary because in a true feedback loop, the effects are heard proportionally to the level of the signal. As the level drops off, the signal returns to a sine wave. Scaling amounts should be 100 for KeyNote and 91 for ModEnv. I set MG D 65.3 ms. Osc1 amount is 44. While this works nicely, my ears want to hear a little bit richer sound than a simple sine wave. Change the osc1 mod to 3, amt 4. I can use FM amt to enhance this attack, so route MG to osc1 FM amt, amount 47. That will give you a nice percussion sound.

So Osc1 will mod Osc2 FM Frequency--in this case there's no way around it because of the feedback loop. So what we need to do first is make sure that Osc2 already has the kind of sound we want. It needs to have a nice acoustic guitar sound--a bit on the mellow side. Set the ratio 3:1 for that classic FM guitar sound. You could route MG to Osc2 FM amt, but I don't think that's necessary--and not really what I'm going for. There is a difference between Op B and Op C envelopes, but not enough for me to be concerned. The FM amount should be just enough to work, so I set this for 8.

Now route Osc1 to Osc2 FM frequency. This could be scaled with KeyNote and/or Mod envelope. This time, however, I want to take advantage of the pitch instability caused by routing oscillators to each other. Simply turn up the amount to 25.

We want something similar to happen in our second stack, except this time we want a more "core" harp sound.

First we need a 4th oscillator. Route osc. 2 to filter 2. Additionally you can route osc. 1 here as well. Here's the tricky part: The filter has to be tuned an octave+5th above the fundamental. I start this by tuning to the fundamental and working up from there. The following settings should be automatic by now: Cutoff 260 Hz, Res 127, env/vel 0, kbd 127. Now, just by doing simple math and multiplying our fundamental frequency 260 Hz by 3, you should have the correct tuning frequency for the filter: 780 Hz. The problem now is that Thor doesn't allow tuning this precise. One click too low sounds grossly out of tune while one click too high sounds like an overused chorus effect. We don't have much of a choice, do we? I say go with the higher frequency: 815 Hz. We have to take some comfort in that this is only a small component of our sound. Everything is important, of course, no matter how small or subtle. And since detuning is a big part of this patch, this one "flaw" might actually help us.

Route Filter 2 to Osc. 3 FM Frequency, amount 11, scale with MG, amount 54. The idea is balancing the modulation effect with scaling to produce a subtle percussive brightness to the sound. Ideally this would sound a bit more like a xylophone, but all we have is all we have. Osc3 is 2:1. Set the FM amount to 2. Route MG to osc. 3 fm amount, amount 50.

At this point we're really finished. I made a few tweaks here and there. For one, I have all 3 oscillators routed to filter 1, balance favoring oscillator 2, and 1+2 level way down in relation to osc. 3. I have a LPF on filter 1 to take a little of the edge off the sound. I want to be true to the harp timbre, so a lot of added chorus isn't going to help anything.

So there you go! Another patch using a 4th oscillator and some very subtle routings. Enjoy!

Would You like some Cheese With That Whine? Adding a 4th oscillator in Thor with E.Organ 5

Previously in the Celeste patch, we see how taking it easy and using plain sine waves every now and then can result in very open-ended kinds of patches than can be hybridized or repurposed for a variety of musical applications. While we generally want to stay away from sine waves, they are still useful for additive sounds in which we aren't concerned much with timbre-variation via filters. Most often these will crop up in electromechanical organ sounds--the most notable of these being the hammond B3. Today, however, will be a slight departure from what we expect from organ sounds since we'll be exploring the cheesy/psychedelic timbre of the Farfisa.

No patch collection is complete without an emulation of the Farf. You'll need this for the 60's organ sound and New Wave styles. Proceed with caution if playing this live, though. It's really cheesy, and your band members may beat you if overused!

Before I forget, I should probably publish my "default" patches. I always start with one of these, and it saves so much time since i don't have to clear out default values or individually load oscillators, etc. My amp EG's don't stray far, and sometimes I don't even bother with setting AEG. I'll save those details for another blog, but you should always have a default patch for different instrument types. I also find that certain instruments, especially the ones I'm working on now, become templates for other sounds. Much of my collection will end up being variations and variations of variations of patches that appear in my blog.

I digress. I have my default sustained instrument patch loaded and ready to go. We all know what a Farf sounds like. Looking at FM8's FM Matrix, we can see ops A/B and C/D are the only mod/carrier pairs. E and F are standalone sine oscillators and are liable to create problems for us when we get close to finishing the patch. For now, let's start with the easy, obvious stuff.

Setup osc. 1 with a 1:3 ratio. The original patch has a feedback loop, so let's effect this in the router. Routing to FM frequency causes for instability than we want, so route Osc. 1 to Osc 1 pitch instead. This sound should have a slightly nasal but full reedy sound. Set FM amt to 8. In the router, set amt to 24 and scale with key note, amount 56.

Now we have a basic organ sound. Let's cheese it up. Route Osc. 2 to filter 1. Set ratio 3:3. This sound has to be really bright to work, so set FM amt. to 79. In the mixer, adjust the 1:2 balance to slightly favor osc2, about 78.

You're almost there. You have some really high, bright partials but not enough fundamental. You need an octave and a sub oscillator. But there's a problem: You only have one oscillator left. So how do you fix this? You can compromise one for the other--say, you think the sub osc. is more important or you'd rather keep the octave. You could use one as a modulator and the other as carrier, experiment with different setups this way until you come up with a sound you like. This is fine except you have to consider that, while you aregetting sum-and-difference signals of both, you're still faced with the problem that one is still modulating the other, hence altering the timbre--however slight. If you can, you need to find another oscillator. It can be done in one instance of Thor.

Activate Osc. 3 routed to filter 1 bypassed. Also route it to filter 2 and plug in a LPF. Make sure the filter signals are running parallel and you have output from both. Change filter type to Type I. Set res. all the way up to self-oscillate. Back off the env. and vel. all the way to 0. Turn keyboard scaling all the way up. Now you have your 4th sine wave. All you have to do now is make sure your cutoff frequency is 2 octaves about the oscillator frequency. Change the Osc. 3 octave to 3. I find the proper cutoff frequency by first matching filter cutoff to the osc frequency, which I find at 130 Hz. That's not frequency you actually hear, of course, because keyboard scaling is all the way up. Now adjust for 2 octaves. By doing simple math, you find this occurs at 520 Hz, except the closest you can get is 521 Hz. This is actually ok because we want to allow for some degree of anomalous tuning--makes the timbre a little more natural-sounding.

Final step: We've got cheese, but not quite the kind you can smell. We really stink it up by adding obscene amounts of vibrato. Route LFO1 to Osc 1, 2, and 3 pitch with amounts 22, 22, and 19 respectively.

One think I like about this patch is how it actually sounds better without effects. I firmly believe in avoiding effects if you can get results inside the tone generation, such as detuning you find in so many DX7 patches. This patch could also go the way of so many B3 patches in which you'd want a rotor or chorus effect. You certainly don't want to overuse this patch. But every now and then, feel free to cheese it up!

Takin' It Easy: Celeste

One thing I'd like to say as an addendum to my previous post regarding the look and feel of your setup is that one thing you really can't live without is a trackball. That's the best invention since bottled beer, lemme tell ya. Mine is a Logitec, the one with the big marble in the middle of it. It's one of those that's been made symmetrically so lefties like myself won't be uncomfortable for all those hours of never-ending tweak sessions. It is a beautiful piece of work. I think the only way it could be better is if they made a wireless version.

Seriously, if you spend more time moving virtual knobs and sliders, you need one of these.

OK, I promise no more pimpin' Logitec, but it's a great product.

There was something else I forgot throughout my previous postings. Yes, I do have a small number of go-to patches that save a lot of prep work. I also have a few variations of patches I make along the way. What I forgot was that I was working through my collection of patches alphabetically before my computer died. While I'm on my backup machine, I'm going alphabetically from ROM to ROM. I was pleasantly pleased, considering my goal for today, that more than half of my work was already done! Not wanting to break momentum, of course, I'll push on if I still have the energy and the time.

The title of this post is "Takin' It Easy," and here's I'm going to discuss some of the lazier ideas of the task of making keyboards. Michael Bierylo in his sound design course will tell you to stay away from sine waves. It's not that sine waves are evil, necessarily. It's just that one sine wave here or there won't really do anything. There are some things you CAN do, of course. For example, you can add a sine wave to a saw wave to emphasize certain partials. Or you can add a sine wave to a triangle wave, several partials up, of course, and detune it for some inharmonic effects. But you won't hear a sine wave if it's the fundamental. Using it as a sub oscillator can be nice, but it still doesn't speak as well as other waveforms.

Add a bunch of them together and you can get some interesting sounds. Or use them as FM oscillators as in DX7 and Thor. But one lonely sine wave?

It appears that's what we're working with here.

Let's start with FM8, ops A and B. There's a feedback loop here--no big surprise, right? Hey, if it were easy, everyone would do it! There's only a tiny bit of feedback and FM going on. In fact, just on computer speakers I can barely hear any sound at all from these two. Let's try something different. Instead of going for some recondite frequency-shifting modulation from feeding an FM pair into itself, let's use the shaper to add just a hint of sine wave distortion. Activate osc 1 into filter 1, leave it bypassed, activate the shaper. Add in a little drive, say, 42. Osc 1+2 level should read -3.5 dB. Activate the left arrow pointing towards filter 2, but don't activate it's arrow. You'll see why in a minute. In Osc1, set a 5:1 ratio (remember, modulator first, then carrier, backwards from what is displayed), leave FM amount all the way down.

We shouldn't really hear anything at this point. That's a good thing since what we'll be doing depends on signal routing. In the router, route amp EG to Osc1 FM amt, amount 27. Route the shaper, amount 59, to Amp Input, scaled with mod EG amount 85. Set MG D to 29.7 ms. This will give you a nice, sharp attack that is barely audible. It is a subtle part of the sound, and if I were going for authenticity, it wouldn't be audible at all. For whatever reason, it's there, and I feel like an effect is no good if it can't be heard. Just my opinion! I'm adjusting my amp EG settings: D 1.24s and same for R.

It's the next part one might find difficult to digest. From the default setting for Osc2, turn FM amount to 0. I want to avoid a sharp attack for this timbre, but I don't want to sacrifice the percussive attack of Osc1. In the router, route Osc2, amt. 82, to Amp Input, scale amount 100 with Filter Env. I recommend leaving S all the way up, although you could turn it off and use a long decay.

You're as good as done at this point. I noticed in the original there's some slight chorusing. Just turn on the chorus, delay 0, F.Back 0, rate 1.21 Hz, amt. 22, dry/wet 42.

Sine waves aside, what else is unusual about this? The chorus takes over the role of detuned oscillators, so there's no need to program another oscillator. The DX7 programming on this is for the most part useless. We've freed up Osc. 3, which is perfect if we ever want to come back to this patch and add more complexity, as in a layered type of instrument. And since all we have is a subtle bell percussion sound, the sines in this patch aren't going to get in the way of other timbres. Another possibility is to get rid of Osc. 2 and have 2 hybrid kinds of sounds in this patch for even greater complexity and versatility.

It's good to take it easy every once in a while. Easy patches like this are versatile and good source programs for a variety of different applications.

32nd DX7 patch

I've just reached my 32nd DX7 patch named "Voice I." That brings my total number of Thor FM patches to 70. W00h00!!! I mentioned in an earlier blog that my number include 7 patches that are only building blocks for others, so that really means I have 63 that are somewhat useful. You can see, of course, that roughly half of those are variations of others while a few others were "discovered" along the way.

This is a big milestone for me as a beginner sound designer on what I hope will turn out to be a huge project when finished. For this post, I'm not going to do a walkthrough of a patch I find noteworthy--I'm just going to take some time to write some ideas I've had over the past two weeks.

One thing I think is important is having a goal. My goal is simple, of course: to build a comprehensive library of sounds using Reason and Absynth along with a few other synthesizers. To reach that goal, I need small, intermediate steps. My current level is just learning the software, specifically Thor. In the process I'm building my skills using FM synthesis to rebuild DX7 patches in Thor. I don't know the next step yet, but there's still plenty of time!

Another thing I think is important is writing things down, like this blog. I enjoy reading the SCI email list, and there's a debate that comes up from time to time about handwriting compositions vs. doing all work at the computer. Personally, I prefer writing all my music at the computer. I understand what they're trying to do, but that doesn't mean I'm into it! Besides, my handwriting ability is poor. I thought about that for a while, though, and I got inspired to do something similar: I'm keeping a sound design journal. That's right. I'm writing (by hand) in a little notebook describing the work I'm doing at any given point in time. It looks almost like a recipe book, and I'll try to describe what I'm doing.

When I start work on a patch, I write the date at the top of the page and the name of the program. Underneath that, I draw a diagram of the DX7 algorithm (derived from FM8. I don't exactly have the algorithms memorized, but I do try to reconstruct the algorithm to look more like the DX7 diagrams rather than the FM8 version). In the diagram, I write the ratios and FM amounts and indicate which ops have feedback loops and the amount. I prefer FM8's values for some parameters because they're a little more precise than DX7.

Next I draw a sort of key that relates operators or op chains to their actual function in the timbre. If something is a percussion attack, or if it's a pad within a more complex patch, or if it's a center frequency in a detuned chorus effect scheme, I write that down.

If I feel the need, I make some comments about the FM8 patch. That helps me solidify what I want to do when recreating it in Thor.

Then I'll make some notes on how I'll make 2 or 3 core sounds that make up the Thor variant.

Before I write anything else down, I go oscillator by oscillator, EG by EG, filter by filter, line by line in the router to find what it is I need to do to accomplish the previous step. Because of the differences between the DX7 and Thor, I can't set parameters in stone until I actually hear what they're going to do. The only part that corresponds exactly to the DX7 is the mod:carrier ratio. All of these have to be tweaked because DX7 patches almost always have ops detuned for chorusing effects. You can't do this in Thor. You compensate for it by detuning entire oscillators. When I get the results I want, I write the settings down. In my notebook, this happens in a semi-random way. My notebook routings don't appear in the same order as in the Thor UI. But they are good enough that I could copy patch settings straight from my notebook into Thor and get identical results.

I don't generally pay much attention to Amp EG settings because my default patches already have those programmed. If I need a pad, I load a pad and forget all about tweaking the amp EG. This becomes necessary when the amp EG is used to scale or mod something, but generally this is one of the parameters I leave out of my notes.

When I'm done listing my settings, I make a few comments on the sound, how it stands up to the original (if at all), and what it can be used for.

So basically that's my patch notebook. The best part is having a reference when I'm about to change another parameter. I don't have to worry about forgetting what I just did or what the next step should be. It also shortens the time it takes to create new patches. All I do is invest a few seconds here and there on writing these things down and be amazed at how easy making patches seems to be.

Anyone who reads this should consider doing something similar. Along the way you'll discover tricks other sound designers came up with to make a certain sound or you'll stumble on something accidentally that you'll want to use later. Write those tricks down, whether yours or others, and you'll find you have a valuable reference when those elements are needed.

OK, aside from keeping a notebook, I had another idea that I put into practice yesterday. I invested in an external floppy drive and a box of 10 floppies. Don't laugh, I know how archaic that seems, but hear me out.

I have a deep love for vintage equipment. I love it, but I just don't own it! So I'm stuck with a DX7 and an Alpha Juno until I can financially justify getting new things in. Floppy drives were common on older instruments although newer instruments have internal hard drives and slots for different memory cards. For me it's the image of the once-ubiquitous floppy drive that contributed to (or detracted from, either way) the look and feel of older synths and synth workstations.

My thinking is that the look and feel of equipment, no matter how irrelevant it SHOULD be, does influence in some way the work we do. I want to get in the mindset of working with hardware synthesizers despite my work now being primarily at a laptop. So I'm saving all my patches to floppy disk. I think I'll stop when I fill up all disks that I have at the moment. In the case of Thor, one patch is about 2k. The practical capacity of a floppy is about 1.2MB. So we're talking about at least Thor patches! That may seem like a lot coming from just one programmer, especially if you're a synth programmer with limited time and only program a few patches here and there for very specific purposes. But think about it: I've just finished 32 patches and some variants of them. Let's say, for the sake of argument, I make 10 variations of each. That's 320 patches. So if I go another round of 32 patches, make 10 variations of each, I'll have 640! In that sense, 640 patches aren't really a whole lot. My ultimate goal isn't rebuilding DX7 patches. Anyone can do that. I need to learn the ins and outs of my software and extend that knowledge and skill to creating good patches. Perhaps old-skool FD storage is a good way to set limits on how far to go with the work I'm doing right now. My iBook is starting to FEEL more like a synth workstation with it, anyway!

More to come later today: I want to get through 8 more patches and 1 more blog entry in which I'll discuss making one of those patches.

SFX 2: TRAIN

I want to start today's blog by saying in the process of making good sounds, learning by imitating others' work, something I've had to really work hard at is letting go of my perfectionsism. I may not report on my work EVERY day, but progress is being made. As of this writing I have 66 Thor patches, 7 of which are "default" settings for certain basic types of sounds: Bass, Decay, Ensemble, Lead, Pad, Sustain, and Main Blank Patch (sus transient only). A few of those are variations. For example, I found in the process of creating a banjo patch I'd made something that sounds very much like a cello. That spun off a number of related cello patches. Although not part of this blog topic, I'm also creating some pipe organ patches for the Korg T2 at my church--needed occasionally when the real pipe organ blows a fuse or is otherwise temporarily of unsound operation (take that as you will!). Michael Bierylo (sound design instructor, berkleemusic.com) is fond of saying we should be deliberate about creating the sound we want, not rely on "happy accidents" to achieve interesting sounds. Although I completely agree, we can't deny that these "happy accidents" do happen. I was working on the DX7 Orchestra patch, which is a string/brass combination when I completely screwed it up. The end result was something akin to a big-band brass section that can certainly be useful one day. While I didn't achieve the perfect "orchestra" sound, I'm convinced it would have been a horrible mistake to throw this patch out.

Part of the inspiration behind this blog is the idea that synth sounds can be recreated in a variety of ways. The final project involved reorchestrating classmates' projects. One of my classmates used samples for almost the entire Reason project. I couldn't resist. using Thor and Subtractor, I reconstructed his sampled instruments from scratch. You know what else I did? I saved the patches I created! They might come in handy one day.

This kind of work gets really frustrating really fast when trying to get that EXACT sound when it's really not possible, at least not without some extension of the method being used. For example, I was working on some DX7 strings patches that rely heavily on feedback loops to get an almost analog saw waveform. The original DX7 patches are just gorgeous. But without those feedback loops in Thor, the sound is very nasal and flat sounding. I spent hours trying everythging I could think of to get that sound, even reverting to making three less complex versions of the basic string sound and trying to combine them in different ways to get something more accurate. Nothing worked at all. So after taking something for my headache, I broke down and inserted a PM synth oscillator. I didn't want to! But sometimes you just have to let these things go.

I think a sound designer is better for it, admitting the limitations of a synthesizer or software. At the very least I can say I know what I'm doing!

Let's get on to the patch of the day. Last post was about an FM sound effect. I'm going to continue with that idea with a little trip to the island of Sodor.

That's right, ladies and gentlemen, we're going to build a train using Thor! I'm dedicating this post to my little boy who, being 2 years old, is hopelessly addicted to Thomas the Train videos (thank you, Santa) and is not a happy camper when his half-hour is up. We're also a bit late with potty training, and not even those fits come close to having to forcefully detach the little man from the TV when it's time for Thomas to go night-night!

Now, you don't really have to have a DX7 or FM8 to follow along with these blogs--I see these writings as more like ways to generate ideas for creating sound, maybe even some helpful hints and tricks to creating other kinds of sounds. But if you DO have FM8, it will be easier to follow along. You can also google Dave Benson's DX7 website, which is about as comprehensive as anything else I've ever seen on that synthesizer and related boards/boxes. If you have FM8 but don't have the patches yet, you can find them on Dave's website.

As always, I'm starting out with FM8, and I've got "TRAIN" loaded. By playing a few keys in the middle of the keyboard, I notice that steam engine sound varies with the key that's pressed. I would think you could do the same thing with an amp or filter EG. Even better, you can sync the steam engine sound to song tempo.

Playing lower notes brings in the "choo-choo" whistle. It's exactly the kind of sound you'd expect on a toy or model train set. Playing higher on the keyboard brings in a train bell not unlike what you might hear at a railroad crossing.

So there's a lot going on for one patch. Very complex indeed compared to what we're used to, but there are a few things about this sound we need to pay attention to.

The main thing is there are three completely different timbres going. Pop quiz: How many operators does the DX7 have? 6. Very good. Now, how many distinctive sounds do we hear in TRAIN? 3. OK. Do the math: 3 different sounds/6 operators=1 sound for every 2 operators. This is perfect for Thor because it's the best effective setup we have.

There's the regularity of the steam engine, which will either be controlled by the note, by the LFO, or LFO controlled by keyboard scaling. It's really up to the programmer how this is going to happen. Personally, I like the most rhythmic or cyclical sounds to line up with the song tempo--that's the route I'm going.

Let's take a look at the guts of the patch now that we've done our listening.

By turning off Ops D and F I can hear only the steam component of this sound. Without the distraction of the whistle and the bell, I can fully appreciate what's happening here. In the DX7 world, there's not much difference between a steam engine and a helicopter! As a helicopter, low notes get slow rotor, high notes make a fast rotor. If the rate is dependent on the note played, you might also get some interesting effects playing with pitch bend, or even use MG to increase the rotor speed over a specified period of time. See how easily a variation of this sound could be made to sound like another effect entirely? Let's start with this sound.

We can easily make a broadband white noise effect simply by switching to a noise generator. If we did that, we can also generate the tone created by the FM pair simply by using BPF with high resonance. My attitude is that after letting go with my perfection streak, I don't really care if it gets preserved or not. I also want to stay true to the FM generation of this sound, so I'm sticking with FM pairs in Thors.

I'm going to set Osc 1 exactly the way it is in FM8 with a 5:9 ratio. There are some interesting things here. First, DX7 is using one of those pesky feedback loops, but it gets better: The feedback loop is set all the way to 100. Why is this important? Because this is how the DX7 generates digital noise. Not that I really care, but for the sake of experimentation, I'd like to attempt to preserve the tone present in this sound. Thor was designed for wider, more general application than some of the more specific things that the DX7 does, so FM amount and routing Oscillators to themselves has more extreme effects on the sound than the original DX7. The tone is high pitched as you'd expect with a carrier ratio of 9. Turning the FM amount all the way jup does change the timbre, but it's a brassier sound instead of the subtle whine we're really going for. Even though it doesn't quite work at every level, I'm ok with a FM amount of about 50.

Now for the feedback loop. Again, this isn't an exact science and the results are not what we expect from the DX7. If you've been following this blog, you'll already know how much time I spend in the matrix router. Route Osc1 to Osc1 FM Frequency. I typically test out my sounds low on the keyboard, but what's interesting here is that the upper frequencies aren't as badly affected at low modulation amounts. While holding the note C5, I'm changing the FM amount and the Osc1FM amount in the router. At an FM amount of 18 and router amount of 31, I get a fairly stable tone. Not quite the FM whine I'm going for, but could be nice for some pitched steam effects in another patch. By increasing the router amount to 37, I have some colored wind effects. As I said earlier, I don't really care that much about this kind of sound, so I increase the router amount to 59. There's still a hint of color, but there's definitely a broader spectrum of sidebands than we started with. This is perfect for what we're trying to do.

We need to rhythmically modulate this sound. Load the LPF in filter 1. Move all settings except drive to 0. I want the tempo sync this sound, a slight modification of the original patch. Set LFO1 to Tempo Sync. I'm tempted to use a down-ramp (using a negative value in the router), but I'm afraid that's going to be too hard-sounding. Instead, I'm going to use waveform 8 because I think a triangle wave would be too soft. This also has the advantage that some DC will still get through. A steam engine is always noisy, so it's better not to operate this LFO as an on/off switch. In the router, select LFO1 as source, Filt1Freq as destination, amount 100. Perfect!

Next let's attack the bell/crossing signal. What's interesting about this timbre? Well, it's a seriously inharmonic kind of sound. What else? Every note I play is exactly the same pitch. Enable Osc 2 in Filter 2 and route the signal into the amp. Turn KBD all the way off in the oscillator. What do we notice in FM8? We have a 977.24:371.54 ratio. Harmonic relationships are very important to the Thor synthesizer, even if we don't want/need to use them. So how do we find the relationship between these two frequencies? Quite simply divide the highest number by the lowest because harmonic relationships are always multiples of each other. The result is roughly 2.6, only slightly higher than an octave+3rd. What are some ratios that give similar results? 3:8, 4:11 (interesting), 5:13, 6:15, 6:16, 7:18, 8:20. Any of these can work, of course, but I'm going with 5:13 or, more correctly mod: carrier 13:5. On a side note, I've never understood why FM synthesizers read carrier-modulator. On the DX7 et al, operators read top-bottom, with higher operators in the chain being the modulators and bottom operators the carrier. The signal chain of synthesizers is typically left-right, so why can't the modulator be on the left and the carrier on the right? It would make more sense. Anyway, I've got Oct2, semi 7, and tune 27 (the last two to get closer to the DX7 in pitch, although I'm not going for absolute mathematical precision here). Now, unlike the DX7 patch, I want to make this bell sound periodic, like the steam engine sound. I still want it tempo synced like the steam engine. it should be at a regular steady rate, but at the same time I want it to sound random. And yet I don't want it random like I'm TRYING to sound random... I don't know... Let's do this: Set LFO2 for key sync, tempo sync, waveform 3, rate 3/8. Route LFO2 to Osc2FM amount. Set the osc2 FM amount to 40 and -69 in the router. The illusion of movement would be nice here, so let's do a crescendo/decrescendo using the MG, all synced to tempo. Set Mod Env Tempo Sync, Delay 4/4, Attac k 4/4, Decay 2/1, Release 3/4.

We're about to run into a train-wreck. We still don't have a way to control the whistle blow (it doesn't exist yet, but it will). Using the second filter would be good for the passing-by effect we're going to create. You could also mod filter 2 drive, mixer 1-2 balance, or switch to osc 3 and mod osc 3 level. The only trouble is once those destinations are gone, they're gone, and we may need to use them later.

How about this: Route Osc2 to Amp Input, amount 100, scale with Mod Env, amount 100.

We need to make a whistle now. On the DX7, there's some keyboard scaling that keeps the whistle isolated to the lower part of the keyboard. At the moment, I'm not jsure how to do key scaling for 3 oscillators in Thor, but for now I'm not really all that interested: There's another way to do this. For now we'll concentrate on the basic whistle timbre.

In this case, operators E and F do have a rational relationship even if it is offset to sound inharmonic. The upside is at least the pitch can be altered by pressing different keys. We'll start with the obvious: The ratio is about 3:1.6. Low to high, that's in the neighborhood of a minor 6th above fundamental and octave+5th, so something like a major 7th will get us what we want. Set osc 3 carrier to 13, mod to 24, oct 2, FM amount 18. Enable Osc3 to filter 2.

We want to trigger the whistle any time we want, but not for every single key on the keyboard. By messing around with the filter, I find I can apply a pseudo-keyboard scaling that almost works except for some very unpleasant (and unwanted) high pitches in the sound if I try to trigger the steam engine/warning bell using only high notes. If I lower the cutoff frequency, I also lower the volume level of the whistle sound I want to keep. I've tried various kinds of scaling and nothing seems to work. I'm not sure how to do a proper keyboard scaling or split using Thor, so I'n at a loss as for what to do.

There is another option, however. In the router, select Button 1 (from Modifiers) and route to Osc3 Level. make sure Osc3 Level is all the way down. Go to the global settings and select a note you'd like to use for the train whistle. I chose C3. I also switched the keyboard mode to mono legato so no matter what the other key I press, it won't interrupt the steam engine sound. Since I'll only use a single key for the whistle, there's no real point in setting the oscillator 3 keyboard scale to full. I cut it all the way down and set Oct to 3. This gives the perfect Thomas-style train whistle!

For Filter 2 (where osc3 is going), I set the drive all the way up, all modifiers to 0, cutoff to 3.62 kHz, and res slightly up to 22.

Finally, effects. I try to avoid a lot of effects unless absolutely necessary, like when I run out of oscillators and all FM8 is doing is doubling up on detuned operators. But to get the right kind of CHICKa-chicka sound of a steam locomotive, you really need a touch of delay. I kept this really simple. Time is 78.7 ms, feedback 25, no modulation, and effect level 36.

In conclusion, you can never have enough effects. Even if you aren't good at programming them or all you're interested in are pitched synth sounds and classic kinds of synth effectgs, you'll do well to try some more advanced tricks like this one. TRAIN is a very complex sound for a single instance of a synthesizer, and these kind of sounds are great for coming up with ideas and inspiration for other, similar kinds of sounds.

It's a cool sound to try to make. And my son LOVES it!!!

FM SFX: REFS WHISL

Sooner or later, we're going to need to some effects in our library. Tonight I'm looking at (listening to) the DX7 REFS WHISL.

First, let's get one thing straight: Music/sound is about LISTENING. That's why I always hesitate to say that I'm "looking" at a patch. I think we have to be careful that we listen because of all the parameters, and I mean hard data, that's right there in front of us on the UI. I'd love to live in a world where there are no NUMBERS on our synths. I also dig rotary controls--no upper/lower limit, we just keep on rollin'. ANYTHING, any opportunity to just listen is the heart and soul of music.

I teach piano lessons privately. Young and old, I tell them the same thing: The stuff you see in front of you is nothing but ink and paper. Music is what you hear. It's communicating in some kind of meaningful way, even if the meaning can't quite be put into words. I think of music more often as that which can't be said. It's a lot of things to a lot of people. I just tend to dwell on the more abstract.

I just hate looking at sound and music as just a bunch of numbers. Unfortunately, the GUIs often communicate with us in numerical form. It doesn't help that the first synths seem to have been developed by mathematicians and scientists. There are some advantages, though, because it allows us to converse on the topic in concrete terms.

Back to the patch:

I'm looking at (yes, looking--sorry) REFS WHISL in FM8. What stands out about this patch? If I listen to one note--well, it sounds like what you'd expect. Play another note, and--whoa... Exactly the same. OK, now actually looking at it, I can see that it's as I'd guess with no tuning ratios. Everything is in absolute, concrete numbers, which means no absolute, concrete harmonic relationships.

Let's look at Ops A, B, C, and F since this appears to be the most sonically complex part of the patch. Op A is 10Hz, Op B is 1Hz. This looks more like an LFO than anything useful by itself. Now we have to think about what LFOs are used for.

Most often LFOs are good for pitch variation (vibrato, trills, usually controlled with a mod wheel to change depth or width of the variation). LFOs are used for other things, of course, but in beginning sound design, pitch is a good target. So it appears that we're using Ops A and B as LFOs of each other. It's FREQUENCY modulation, right?

Well, there's a slight problem with this line of thinking. While frequency variation, whether with LFO or FM modulators, makes for some interesting timbre variation, FM on the DX7 isn't frequency modulation in the truest sense of the term. There is some variation in pitch, but not really that much. This becomes very evident when using Thor to copy DX7 patches because using one oscillator to mod the pitch of another causes extreme variance in pitch/timbre. There's so much variance that the resultant pitch and sidebands are horribly unstable, which explains why feedback loops in Thor don't really work very well. Why is it FM works so well on the DX7 when it sucks on the more advanced Thor?

The answer is that the DX7 doesn't really change frequency, at least not very much. What happens on the DX7 is that, while there is some apparently detectable change in pitch, it's the resultant change in PHASE that is the true power of DX7 FM. Putting all that energy into altering the phases of sine waves allows for controllable sidebands that add a distinctive coloration to the DX7 timbre. I really admire the DX7 for this because adding just a touch of feedback to an operator or chain of operators gives the DX an almost analog quality. This really comes through on brass patches!

So what does that mean if Thor's brand of FM is pitch unstable when compared to the DX7? Quite simply, if you increase the amount of modulation one oscillator has on another, the resultant pitch is higher than you expect. In a future post, I'll explain how tuning--careful tuning--can compensate for this unfortunate side-effect and get more authentic FM sounds.

Meanwhile, we're going to pretend we don't know any of this and that Thor is exactly the same as the DX7 (you can stop laughing now).

Our first task is to get an FM pair in Thor to behave like not one but--wait for it--two LFOs, one modulated by the other. Can it even be done? "I'm sorry, Dave, I can't do that."

"Open the pod bay door, HAL." Excuse me while I give this thing the finger.

OK, fine. Whatever. But let's go back to FM8 for a sec. Everything else falls within the audible range. All we need is two LFOs, and in doing so we can get very precise results. In Thor, set LFO2 about 10Hz and LFO1 1Hz. In the matrix router, set LFO2 as source and LFO1 Rate as destination. I'm just guessing here, but try 25 for an amount.

Now here's the part where you have to use your brain. Remember, Thor only understands harmonic relationships. Totally guessing again, I'm trying mod 11 and carrier 8 with LFO1 modulating Osc1FM frequency, amt 65 in the router. Pretty wild, right? I'm checking this against FM8 and it's way off. I'm changing the routing to Osc1 pitch and backing the amount to 10. It still doesn't sound like ABC in FM8. What we need now is some envelope modulation. Set the mod EG decay time to 45.2 (keep it short). Remember how we used LFO2 to mod LFO1 rate? Now scale this with the mod EG. Crank it up to 100. Doesn't really do much does it? Set Filter EG decay time to 316 ms and kill the sustain. Use the Filter EG to scale LFO1 modulation of OSC1 Pitch. Set the amount to 100. If you play a lot of rapid notes, it almost sounds like a triangle roll.

Of course, the differences are so striking between Thor and the DX7 that it's impossible to pull this off without stopping and experimenting along the way. I changed the Osc1 carrier to 7 and FM amt. to 22. Getting better.

So far we've only used Osc1. Sounds like a great high-pitched metallic bar kind of sound. A ref's whistle is high pitched, right? Now for the rapid pitch and volume variation associate with a whistle sound.

To my ears, the most significant thing about it is the indeterminant aspect of this sound. Turn off the keyboard scaling for Osc2 and route Osc2 to Osc1 pitch. Make sure FM amount is all the way down, also. Turn the octave all the way down. Here's where the ear kicks in. Important point to remember here is that the DX7 FM effect is very subtle--change phase, not pitch. So for the amount of modulation for Osc2 to Osc1 pitch, set it to 15. The modulation is a bit slow for my taste. Speed it up just slightly by setting Osc2 semi to 6.

Almost there. Repeat the above procedure for Osc3. This time set semi to 2 and amount (in the router) to 39. So close!

The problem we have now is that the bandwidth is fairly broad within our hearing range. This means higher pitches are pushed out of hearing range and the lower end tends to dominate. Now we have to compensate by altering the fundamental pitch of Osc1. Set octave to 4, semi to 9 and tune to -12. Getting closer. Now the trouble is there's too much noise. Here's where filters come in. Set Filter 1 to State Variable, use the band pass filter, freq to 3.28 kHz, res at 123. Not perfect, but it's there.

I also did a little more tweaking in the router. Osc2 amount is 12 and Osc3 amount is 26. Done!

This is a great effect for a lot of different reasons. Obviously, if you NEED a whistle sound, there ya go. But you can get different effect by playing around with the FM amount and mod frequencies in Osc 2 and 3. You can make some great cricket and cicada sounds as well, or play around with the amp envelope to get different kinds of shakers or clackers.

Also keep in mind that part of what we're doing here is a study in how FM works, especially in Thor, and I'm a long way from being an expert. It's a great accomplishment, though. You really can't ignore the possibilities in the software for programming effects. I think it's good practice to work on these, especially given the difficulty of such a task.

Going For The Obvious: E.ORGAN 1

Mastering a new synthesizer is a lot like mastering a new video game. There are all kinds of twists and turns along the way. The bad guys get tougher, and there are usually a lot more of them as you wind your way through whatever virtual world you happen to be in. There are dangers, pitfalls, and careless mistakes that put you right back where you started.

A divergence between synths and video games, though, is in video games each level of mastery can get increasingly complex and difficult. On a synthesizer, a program might arise that is deceptively simple. It feels like you're going back and playing an earlier, easier level of the game. Video RPGs are not often played by going BACK a level, though this is certainly an option. Perhaps you missed one or more objectives. Perhaps there's a magic gold coin that you MUST have to complete another level farther down the road. Or you just want extra points. But typically you aren't working backwards.

On synthesizers, you're going to write easy patches from time to time. But just as with the video game, the level may appear easy, or you made it through on pure luck ONE TIME. What you have to beware is the unexpected demon lurking in the shadows.

That's the discussion for today: Going for the obvious, or when patches look easy.

Let's take a look at E.Organ 1. This is modeled on the Hammon B3 organ. Before we proceed, what is a B3? At the most basic level, it's a tonewheel organ. Everyone knows that, right? Just a bunch of sine waves, right? That's going for the obvious--but it's a deceptively simple process. Saying the B3 was "just a tonewheel organ" is like saying the 1980 Firebird TransAm was "just a car." There are a lot of keyboards out there, just like there are a lot of cars. The difference isn't just the body, it's what's under the hood.

What were some of the things that made the B3 special? Drawbars. Tube circuitry. Key click. Percussion. Chorus/vibrato. Leslie cab with horn rotation adjustable in realtime. Sure. But that's just what you found on the dashboard!

B3's could be really noisy, especially the older they got. What about tonewheel bleedthrough? Aged tubes? What, you really think 50 year old tubes are going to sound the same as the day the B3 rolled out of the factory? I don't know how else to explain it, but electro-mechanical instruments are BAD quirky. If the B3 isn't bad enough, what about those first analog synthesizers that NEVER stay in tune?

I'm not trying to overthink a simple patch--just trying to give a little perspective.

Let me sit down at my DX and load up FM8 on my iBook. I've got E.Organ 1 right in front of me. Oh, and before I forget: I'm looking at the original DX7 presets, not the mkII. I tried loading the originals on the mkII and it was epic fail. The DX7II was an improvement over its predecessor, not a copy with extra features. FM8 does a better job as an analysis tool, but the sound is WAY clean compared to the DX7. So any analysis using advanced tools has to be done with a grain of salt.

And by the way, that reminds me: FM8 really is fairly advanced for what I'm trying to do. As I've said before, Absynth goes a long way to completing the big picture (I don't have it at the moment). But the best way to get a true DX7 timbre analysis is to sample the waveforms while turning various operators on and off.

So anyway, the E.Organ 1. We're using 6 operators in one of the last algorithms on the synth. This means there are NO modulator waves. Yippee!!!!! OK, let's load up two Thors, no FM amount, and copy the harmonic ratios; adjust EGs to get the right effects.

NO.

WRONG.

We're only going to use ONE Thor.

WHAT??? Are you kidding me?

The obvious isn't so obvious anymore, is it? Let's review: What's the goal, a faithful replication of the sound or a close approximation? We don't have to stay true to the original, we're just using it as a model and getting as close as we can. Let's look at patch in FM8 again.

Op A has fast attack/decay time and a harmonic ratio of 3; in other words, an octave+5th. This is our percussion, which is not the most important part of the patch.

Op B, ratio 1, is part of the core sound. So is Op C (ratio 0.5), Op D (1.5), Op E (1), and Op F (0.5). All values are rounded, of course, and do not take chorus detuning into account.

Notice what we have in common: Ops B and E (1), C and F (0.5). We no longer have need for chorus detuning, so we can condense these into two oscillators. We also have Op D, which is basically an octave+5th above our lowest (0.5) operator. Remember, Thor doesn't do anything less than whole numbers, so we need to program twice the value of these essential operators: 3:2:1. Why is this so interesting? This is the additive basis for a saw wave! So let start with that.

In Thor, make sure oscillator 1 is active. By default, Thor's FM oscillators are set close to a saw wave, or at least the FM equivalent. Bring the FM amount down to 4 or 5.

Now, our organ isn't a perfect analog saw wave; remember the B3 was a tonewheel additive instrument using drawbars to control the timbre. On the DX7 patch, all overtones (or DX7 virtual drawbars) have equal treatment. Our job is nearly finished with just one oscillator, but why stop there? Let's active osc. 2 and boost the third partial.

We have some options here. We can use a plain, boring sine wave and set the carrier ratio to 3. This does work, but it's not a very rich, full sound. We could change it to a saw wave by setting the mod ratio to 3 and kick the FM amount to 4. Fine, but we seem to be missing the second partial. Set the carrier to 2 and leave the rest of the settings as-is. What do you think?

So we just rebuilt the DX7 preset E.Organ 1 with only two FM pairs in Thor! Amazingly efficient, right? But we're still not done. Remember the B3 percussion sound in Op A? Activate Osc. 3. Up to this point, we've been setting our ratios twice as high as they should be. So for accuracy, they will all need to be tuned down an octave. Osc. 3 we can leave alone.

Now, the DX7 uses a sine wave for the percussion effect. It's a very short town, however, so there's no real reason why it HAS to remain a sine wave. I took some creative license here and programmed carrier 3 (as you'd expect) and mod 2. It's just a little brighter than what you'd expect, but it's not bad! FM amount is 2.

We're dealing with very small, subtle settings. A B3 is much more complex than what you'd expect, but it's not brutally strange.

Now for the fine tuning: the B3 is a simple on/off switch, so set the amp EG accordingly. Route the mod EG to Osc 3 level in the mixer, turn the Osc 3 level all the way down, set an appropriate amount in the matrix. Set the mod EG decay time to about 1 sec, release at 45.2 ms. In the matrix, set the amount to 88. We could use a good vibrato effect. Route the LFO to Osc 2 pitch, I like amt about 16. I prefer a triangle wave, but a sine will do fine as well. I like the rate a little on the fast side, about 8.47 Hz.

This is all you need, the GOTTA HAVE IT stuff. Of course, a little chorus effect would be nice, and delay if desired. I'm always scared I'm going to overuse these effects. But think about the DX7--every patch I've copied has in some way used detuning to simulate chorus, so there's certainly no shame in relying on chorus.

Now you have a perfectly good E.Organ 1 sound. Enjoy!

Another day, another patch: DX7 Clav 1

My girl turned 1 year yesterday, and amidst the hustle and bustle of the holidays, it was impossible to have a proper birthday party until today. Since "over the river and through the woods to grandma's house we go" is a 2 1/2 hour drive, I thought it a fine opportunity to do some work. I thought now might be a good time to share some specifics on my work. I'm sure to more experienced Reason users what I'm doing is nothing to really celebrate, but it's a big deal to ME! Besides, if there are other users on my same level reading this, it's bound to be helpful.

Here I am working my way through the DX7 factory patches. I decided today I'd recreate the Clav 1 preset. To start, I loaded up my Default FM Decay patch--a very simple 3-osc. FM patch (only the first osc. is active) with an ADR amp envelope. My main problem is that I'm working on the old iBook and can barely see the display for all the sunlight streaming in. I proceed to work over 2 hours trying to figure out how to make one stinking feedback loop and simulate the effect of a pair of carrier waves with two modulators stacked on each in serial with 3 hard-wired FM pairs. Generally speaking, there are a couple of simple ways to approach this.

The #1, absolutely MUST do this step in making a sound is analysis. I load the DX patch in FM8 and switch to spectrum view. To make a long story short, the important thing is switching operators on and off to figure out what effect each operator has on the composite sound. From here, the sound designer has a choice: I can approximate the sound of the combined serial operators by using parallel FM oscillators arranged in pairs. While I've gotten good results from this in the past, certain kinds of instruments are difficult to assemble this way, ESPECIALLY bright, distorted, brassy kinds of sounds. I feel that the results here are still too sinusoidal for what I'm trying to do. DX7 FM sounds have an almost fat, analog quality and evoke the sonic behaviors of acoustic instruments. This happens mainly as a result of how the DX7 uses feedback loops to get waveforms other than sine. Thor allows no such variation. There simply are no means of creating simpler waveforms that make up the basic components of this clavi sound. This clavi is a very dry, pinched sound, maybe a little nasal and definitely a bit on the thin side. I'm thinking a combination of HP and LP filters can make this work (possibly, I don't know yet). Now, another very useful analysis tool is Absynth. It's a good testing ground to better test what happens in FM synthesis by using Transform in the wave window. I only have Absynth on the MBP, so all I have at the moment is FM8 and my ears.

The other choice I have is to use the modulation matrix to do some fancy routing to create the necessary feedback loops and thus some necessary timbre variation (getting closer to that thin nasal sound I mentioned). I decide to go this route.

For roughly two hours, I experiment with a plethora of various mod routings to get the same feedback effect as the DX7. Eventually I had to give up and start over. I was disappointed because I felt I wasted a perfectly good drive when I could have had a great Clav patch. It seems I'd forgotten one of my own guiding principals: This collection is INSPIRED by the DX7, not a direct replication of it. I start to wake up somewhere between Carthage and Yazoo City on the return trip. It was here I figured out what I needed was a compromise between fancy routings vs. keeping it simple and staying true to the original DX7 preset.

Here's what I came up with: Serial arrangements of FM operators are essentially like performing multiplication (it's more complicated than that, I know, but the logic makes sense when you take this to Thor). The ratio of A:B:C is roughly 8:0.5:2. In this case, the first 2 numbers can be multiplied to get this: 4:2 using two operators. Thor does whole numbers only, so perhaps a different way of looking at this arrangement is 16:1:4. Let's say I ignore the first modulator, I get the ratio 1:4. I'd previously experimented with 8 and 16 both as modulators and I found that it sounded to much like emphasizing a high partial completely uncharacteristic of the Clav 1 sound. Now, the 1:4 mod:carrier ratio isn't all that great either, but I'm limited on choices here. I already know I want to modulate FM amount using the filter EG, so I set this up in the matrix. I can't get a true feedback loop, and I certainly don't want the extreme effects of self-modulating a modulator. Instead, I set up Osc1 to modulate its own pitch and set a fairly high amount of FM. The result is not a bad almost-clavi sound. But it's still too much of a filtered analog saw wave to really be useful.

I'm going back to the FM8 patch and having another look at the second set of operators. The ratio is 6:0.5:0.5. I'm thinking we can do something similar as with Osc1, but maybe with two pairs of oscillators this time. To compensate, I'm changing the ratio to 12:1:1 and bringing Osc2 and Osc3 down an octave. Osc2 ratio is 12:1, FM amt 24. I deactivate the output of Osc2 going into the LPF--we won't actually hear this one. I leave the Osc3 ratio 1:1. I route Osc2 in the matrix to Osc3FM frequency and scale it using the filter EG. Getting better! Not quite the DX7 patch. Not even close, honestly. But still a usable clav patch.

Here's where inspiration took over: Listening to a real Hohner D6, you'll notice there's typically an audible click when the damper return to the string. This is a necessary component of the clavi sound that the DX7 is trying to imitate. However, this click isn't present at all in the DX7 patch. What we need here are two separate volume controls via EGs.

First of all, I tweaked the amp EG so that decay is 2.61 sec and release is 122 ms. The release has to be fairly quick, and JUST long enough to make that click audible. Now to create the click effect, the filter EG will have to be inverted. Vel and Kbd amounts will have to be all the way down, envelope amount all the way up, and cutoff fairly high at 8.84 kHz. A typical ADSR setting in the filter EG won't work anymore. Sustain and release times are all the way down, while A=1.02 sec and decay is all the way up. On a basic level, this will work well with your filter, but the filter EG is affecting more than just cutoff. It's affecting also the relationships between oscillator FM amounts and oscillator pitch. Where filter env is controlling Osc1FMamt, I program -47 (remember, we have to invert our envelope to account for that clicking sound on release). Where is scales Osc3FM I set -87. Perfect!

Actually, it sounds more like a cheap imitation of a harpsichord than a true D6. It doesn't even really represent the DX7 preset very accurately. Part of this is due to the fact I'm using rounded-off settings for FM frequencies while the original DX7 is slightly offset. I can compensate by using lfo to vary pitch slightly or I can detune the oscillators (or use chorus). There's definitely a loss of authenticity here between my imitation and the original DX. But it's still a very good clavi sound as well as an open door for other kinds of related timbres.

In conclusion, I'm including an image of this patch so you can get an idea of how it looks.

Inspiration

So it's the new year today, which also means it's my daughter's birthday. I'm still busy setting up my backup computer to get as much done as possible until I get the MBP back. I'm only human, which means I get distracted and sometimes fail to get much done from day to day.

That's OK, though, because the interruptions only serve to help me to reflect on what's really important here. And sometimes those things that are really important are worth dwelling on long enough to keep my goals in perspective.

I'm talking about inspiration, here. Designing sound and composing are things we do because, well, that's what we do. We can't wait for the muse to descend from heaven and whisper in our ears speaking French. Work has to get done. But isn't it great when you get it?

The main thing is the environment in which you compose. Take my environment, for instance. I don't get to spend much time at home until the off-seasons. When I am at home, I'm so focused on composing and sound design that I often fail at paying attention to my family and my living space. This all came to a head as I was keeping my children at home away from daycare and travelled over the holidays to visit friends and relatives. I live in the Mississippi Delta in a double-wide on what was probably at one time a cotton field. In fact, there's farmland just beyond my back yard. The trouble with the cold, rainy season after the fields have been plowed under is that wild animals suddenly find themselves homeless and decide that humans aren't bad roommates.

So it is I find my home infested with mice, which might be tolerable if not for that SMELL. At first we just trapped them. But at this point we've found we can't get rid of them fast enough. We've now resorted to a particularly brutal poison--brutal in the sense it's a slow-acting poison that is cruelly attractive to the animals. My only thoughts this morning were towards thoroughly scouring away the filth in the living room. In place of composing or working on the Yocka Sound project, all I wanted to do was clean house. All day. And rearranging furniture at my wife's request (she feels compelled periodically to do this and today was a good day for it).

The smell improved a little, but the kitchen still reeks. After being completely exhausted from cleaning the living room, I no longer felt motivated to be quite so thorough in the kitchen. We decided to settle for finding any remains that might be the source of so strong a scent. In the process, we frightened one of our uninvited residents which I followed to an unintended entrance. I sealed the entrance (a hole underneath the central AC unit) and resumed my search, satisfied for the moment that there seem to be no more corpses to be found. We'll resume our housecleaning at a better time. For the moment we are elated that things seem to be improving.

Ladies and gentlemen, it's time to clean house! It's time to celebrate our children's birthdays. It's time to spend more time with our loved ones. And in all we do, it's time to find inspiration. Even if it means putting off important work for a little while, we have to find that extra burst of energy or at least clear our channels of energy to let it flow the we way we need it to.

So what does ol' Rho find inspiring? Well, I'm a Southerner. Mississippi, for what it is and isn't, inspires me. The trailer park inspires me, or at least shapes my mood and the creative direction I decide to take.

I've become an avid fan of the "True Blood" series for a variety of reasons. First, I find the deep south setting to be truly fascinating. Bon Temps is representative of small, isolated communities that time forgot. You can drive around the Mississippi Delta and get the same sense of traveling back in time. It's also representative of the kinds of places where uneducated freaks and weirdos are the norm and it's the vampires and other monsters that make up the echelons of high culture and civilization. Equally fascinating (beyond the obvious relevance to my way of life) is the whole pastiche they've created by combining drama, comedy, and soft-porn to make bold, serious AND satirical socio-political statements.

Anyone who knows me well at all can vouch for my disdain for all the vampire craze that has captured the fascination of our youth. I've seen all the Dracula horror flicks and especially "The Lost Boys" (you HAVE to love the 80's!) enough to know that vampires are evil and destructive. I find it unpalatable to accept anything relating to vampires and present day "vampire culture." And yet I find "True Blood" really speaks to me. While I'm vacuuming and mopping the living room floor, I'm passing the time with a "True Blood" first season marathon streaming off the DVR.

Is something wrong with me? I hope not. We don't have to understand those things that put the fire in our bellies. But we're foolish to ignore it. The question here is this: How do we take these things and put them to music? "True Blood" appeals to me in spite of my disgust for anything vampire-related because it reaches my sense of irony and humor. It reminds me that I'm allowed to dwell on the silly and/or grotesque in my own work. Just when I feel that maybe aesthetically I'm headed in the wrong direction, it tells me "Hey, it's OK to be stupid every once in a while." It doesn't have to be about what everyone else is going to like as long as it expresses what you have to say in your work. And while it's trendy to be ironic or retarded these days, we also have the freedom NOT to act like idiots as well. I do believe that there are absolutes in this world--but they don't have a place HERE.

You can make up your own mind about your likes/dislikes. Same thing with sound/music. The key is the generative process that leads you to create. Where does inspiration come from? A silly TV show? An unwelcome guest? Your daughter's first words at one year? That quirky synth SFX you heard on the radio? You can't always wait for it. But it's great when you have it. And it isn't always that far away.