My actual patches

I've been considering making my own patches, the one's I'm blogging about, available for download. I figured a good way to try was to find a file-sharing website. Let's see how this works!

Celeste.thor

Refs Whisl.thor

Toy Piano.thor

Train.thor

Return of Thor

So I'm getting deep into sampling with the Akai S2000 and have made some great progress. It's good to have a lot of things going on to break up the monotony of working in one particular soft synth. I figured out a good trick, something that's easy to implement--keep the old Mac and the sampler at home and ONLY edit Akai programs at home. That doesn't mean I can't prepare new samples while I'm at work. It just means I have to wait until I get home to really work on putting everything together.

So I'm back to programming Thor. Still just one or two patches a day, but I'm moving forward with it. I'll be sure to share a new patch in the next day or so. The good news is I'm slowly grinding towards the end of the DX7 ROM 2a. I really am down to the final stretch!

More on sampling...

This week it seems I've been non-stop sampling with the S2000. I've made a few changes in the way I'm working which makes things faster. But it can also be a pain. Here I'll try to explain as briefly as possible what happened yesterday up until the early hours of this morning.

I started out by getting a high-powered breakfast, my ritual when I've got a big day ahead of me. That consists two pints of milk--one white, one chocolate because I don't like white milk and store-bought chocolate milk is too sweet--and a cinnamon twist from local donut shop Shipley's. Normally I'd prefer a chocolate iced donut, but my wife and I had stopped in too early. I made a pot of coffee when I got home. And let me tell you, there is NOTHING quite like working all day on a project on just a quick morning sugar rush and a steady stream of java. I made 8 cups and slowly sipped about half a cup at a time, roughly once an hour. Something else that helps is drinking LOTS of water.

First step is identify your source and record. For me this is easy: Absynth. I choose this one because it's the easiest way to draw waveforms, and I've built up a nice library of single waves. I use Absynth's record feature to get a 1/4 second snippet of 4 octaves: C0, C2, C4, and C6.

Next step is trim the samples. I've found this works best, even if the note is just slightly out of tune: C0=1348 samples, C2=337, C4=84, and C6=21.

Before I continue, let me say these samples will be out of tune and, for the moment, will have to be tuned manually. Why is that? Notice that the sample length (at 44.1k) of all 4 notes is a whole number. That's because samples are absolute; there are no fractions of samples. I forget the exact frequencies of those notes, but I do know that those frequencies do not fall neatly into any subdivision of the 44.1 kHz sample rate. What this means for us is that each new cycle is going to be a variation of the source sound until enough cycles have passed that it falls back right again. Don't ask me how many time it takes because I don't know, nor do I care. If you know how to use a calculator, you can figure it out. I'm perfectly comfortable tuning samples. Anyway, because those sample lengths are only approximations of the full wave cycle, each sample will tend to be slightly sharp.

Notice that the lengths of C4 and C6 don't fall in the same logical pattern as C0 and C2. Why is that? Notice that C2 is an odd number of samples but directly proportional to C0 4:1 based on a cycle of 1348 samples. Remember, there's no such thing as a fraction of a sample. For C4 and C6 to work, they have to be based on a slightly lower number that can be further divisible: 1344. That way, you can maintain the 4:1 ratio with C4 and C6. But what's the obvious problem here? The tuning will be pushed further sharp, a problem which will have to be corrected later.

So the samples are trimmed. A hard lesson I learned early on was that the S2000 only likes samples of a certain length. I haven't been adventurous enough to find out exactly what that is, but I did learn that 1348 samples seems to be long enough. The S2000 doesn't seem to like looping a full sample, either. So I do know that the sample length for a full wave cycle at C0 has to be at least twice that. I also figured out that the S2000 doesn't necessarily leave the loop points where I set them. Assuming that the points will be shifted somewhat, we need a third wave cycle as a safety. So that's 1348 3 times.

The next step, then, is to copy the wave onto itself to yield an appropriate length, 1348*3 for low notes and 1344*3 for high notes. This is where the samples part ways with the MBP. I load 'em up to a convenient floppy disk and transfer 'em to my almost 10-year-old PC, still running Win98SE perfectly stable! I have an audio editor that, though extremely dated at this point, works wonders with sample information. Now if only it had batch processing... The procedure goes like this: Select all, copy, move cursor to end of sample, hold down paste until the desired length is reached, tab to next window, repeat. This goes very quickly using only shortcut keys, no mouse.

After lengthening the samples, loop points have to be set. Since each sample is the same length and there is no variance in the waveform, the loops can also be all the same length. It's not a hard guess: 1348 for low notes, 1344 for high notes! The difficulty has been that this is a lengthy mouse operation, selecting an exact length and looping to selection. I'm wondering if manually typing in the info would be faster, since it should be the same for all samples.

The next step would not have made sense to me had I not learned a hard lesson last night: My particular sample editor works best in wav format and doesn't seem to output aif properly, at least not in a way that's compatible with the S2000. When I tested the samples, there were not looping. When I checked to see what the problem was, the loop length had been increased by one sample. It worked fine after I jiggled the alpha dial, but I'll leave it to the reader to guess what happens when the loop length gets jacked with! So for the way I work, the output has to be all wav from the Mac to the PC.

After the WAV files (NOT aif) have been trimmed, lengthened, looped, and root keys set, they have to be saved in a format friendly to the Akai, which IS aif. My earlier mistake had been trying to work all in aif in order to cut out a few steps which shouldn't be necessary (they are). I herd all my growed-up samples back onto floppies for the drive back to the MBP. The floppies are then erased (again, for the third time, by the way) after the wavs have been transferred to the Mac. I fire up Logic, load all samples into the bin, and then batch convert the whole lot of 'em to aif.

These go BACK to the floppies which are then loaded into the old PPC Power Mac and copied to a desktop folder. Mesa is already up and running, so it's a simple drag-and-drop into the sampler memory. Finish line!

For now, I'm fine tuning all samples from the face of the sampler. This works for now because all samples have a strong fundamental. The trouble will be when sidebands cause enough trouble to confuse the tuner or when additive waveforms emphasize an overtone which will be detected as being out-of-tune even if it isn't. In the future, fine-tuning information will have to be included in the wav file. It will certainly be less time-consuming.

Beyond that, there are a few nit-picky items to handle. For example, this process leaves the file extension in place. I like to rename all samples so that they don't have the extension. It takes a little time, but MESA makes that much easier.

Here's a wrap-up of the entire procedure:

1. Record samples with Absynth, save in wav format.

2. Trim waveforms in Logic.

*Cycle lengths are (in samples): C0=1348, C2=337, C4=84, C6=21

3. Transfer to PC

4. Copy/Paste samples to proper length

5. Set loop. Length is 1348 samples for low notes, 1344 for high notes.

6. Transfer back to MBP, load into Logic.

7. Batch-convert to aif.

8. Transfer to Mac PPC.

9. Use MESA to transfer to S2000.

How to make small samples that WORK

Let me start by telling where I am with the Thor project. Right now I'm having a lot of trouble staying motivated and inspired to work. I've taken some time off from the Thor project. While it was refreshing, I've also found that I'm getting lazy. It's very difficult to stay focussed. At this point, I'm feeling like working on that project is a chore.

It's the BRICK WALL. We all hit it at different times in various projects. It's not a good feeling. So how do we get through it?

The answer comes from advice I give my piano students: Proceed slowly. Getting in a hurry causes us to slip up, make mistakes, overlook important details, and form bad habits. Some brick walls are easily knocked down with a big hammer. But what if we lack the strength/discipline to wield that hammer or the wall is just too thick? Easy. Just get a smaller, lighter hammer and a chisel. SLOW DOWN. Conserve your strength. What you find is that the creative ability is strengthened through slow, deliberate work. The bricks in my wall are called Distraction and the mortar is Fatigue.

I moved my S2000 to my evening office. That went a long way to bustin' up some bricks. But that does nothing for feeling mentally, creatively, and even emotionally worn out. No biggy.... Just slow down. And that leads to the next piece of advice I give my frustrated piano students: Slow progress DOES NOT EQUAL no progress. It takes time to develop a skill in music. Chip away. Having said that, I might have a lot of time during the day, but I'm reserving that for only one or two patches a day and not the four to eight I've done in the past. Perhaps there will be a big sprint to the finish line, but it will have to wait until I can chip away at some more of this brick and mortar. We'll make it!

What I really want to write about are a few little discoveries I've made working with the Akai S2000.

I started out creating waveforms in Absynth that ranged from silly simple, like saw waves, triangle waves, pulse waves, to complex FM variations of those as well as "classic" FM (sine operator modulating up to four additive waveforms) and simple sinusoids. My next task was transferring them to the Akai. This was no big deal. I took the shortest sample I could get away with and attempted to loop a single cycle. I also thought it would be a good idea to leave off the anti-aliasing filter for a good "digital" sound. After spending many days on this, I thought I had a good things going.

Apparently I thought wrong. When I began assembling these first attempts, there were pops everywhere, and I didn't even want to think about how long it would take to fix the problem. I made it a point to stay away from the sampler for about a week because I knew there was no end to that frustration. I was also having a lot of trouble keeping my mind on Thor. Ultimately, this led to moving the sampler to my other office late last week.

Here's what I think happened: Absynth (hard lesson learned here) does not output a consistent waveform. I checked. I turned the unison random setting all the way to 0. It really made no difference. That makes it impossible to find a decent loop point consistently.

The other problem had to do with aliasing. Because of all the digital noise introduced in the upper registers, it's really impossible to find the wave cycle. And if you can't find the wave cycle, it's impossible to set the right loop point.

I discovered all this purely by listening. I can't stress that enough. When my ears could no longer explain what they were hearing, I had to resort to a visual editor to see if I could figure out the problem or confirm my suspicions.

What I found with my eyes was exactly what my ears were telling me: Absynth is horribly inconsistent. I have some theories. I'm running Absynth at the standard 44.1 kHz. My guess is that running up against the Nyquist frequency is causing some phasing problems. It could be that would happen anyway, though. If Absynth uses dithering to handle noise with anti-aliasing turned on, there should consistently be some differences in the waveform from cycle to cycle. Again, this makes looping a single cycle impractical. To check waveforms, I used Absynth to record itself (standalone version) and ported the results to Logic for trimming. It became a contest to see which cycle ought to be sampled, and it's not to be won without difficulty.

The logical conclusion is that I'd have to resort to a visual editor (Logic) to pick out these single cycles and loop the full sample. That takes all the guesswork out of setting loop points and should, in theory, render it a non-issue. Away from the Akai, I decided to test this theory in NN-XT. The result was flawless! I was making some great pads within seconds.

After getting my method down, I tested this theory on the Akai. This time I took an old Dell Inspiron with software capable of sending SDS and MIDI'd it up to the Akai (still don't have SCSI yet). The sample transfer felt almost instantaneous. At first, the loop sounded automatic (it was a saw wave). It was good like a good loop should. But the higher samples didn't sound right. So I scroll through Akai pages to try to get to the bottom of the loop problem. I got to the sample trim and the Akai got totally confused. The sample was smaller than what the Akai was prepared for, so it stopped working entirely. Complete failure. The workaround is to use the group buttons to scroll past the trim pages. After accidentally causing a few more failures, I got to the sample loop page and found there was very little room to adjust. I'm not sure why, but Akai just refuses to simply loop an entire sample, especially one that small. Am I missing something here?

Now, one mistake I made that I won't repeat was using saw waves. The problem with saw waves is they inherently have that "pop" when the cycle restarts. That makes them easy to tune. Another side benefit is looping various numbers of saw cycles for a syncing effect. This will introduce more harmonics or sub-harmonics (and they say audible loop points are a bad thing--bah!). The down side is this really only works for saw and pulse waves. While this CAN be useful for other kinds of waveforms, I generally advise against it. But the point is that tuning the "perfect" saw wave is impossible because you can't really tell whether the wave is "syncing-up" with the loop point or if it's the beginning of the cycle. Another analogy is to visualize a physical ramp. If you were to cut off the upper portion of a ramp, you'd just have a shorter ramp. If you cut the ramp in half, you'd still have a ramp. Same phase, same angle. Due to this fact, every point on the saw ramp is potentially a loop point. Good luck actually finding the right one. Using a tuner will help you get closer, but you still have to wonder if you got it exactly right.

So here are some observations: The samples have to be longer. Let's say that we make the lowest note we sample (C0) a point of reference. My ideal would be to make it 3x longer and loop the 3rd cycle. Easy. Next, we need to make all higher, shorter samples equally as long in terms of time and number of samples as the lowest sampled note. Also easily done. Finally, every wave cycle must be identical to the first. By fixing the first two problems with a simple cut-and-paste operation, each successive waveform WILL be identical.

Hold on, we're not done yet! Because of problems with Absynth not producing a consistent waveform, we can't simple relegate loop points and length to a simple function of proportion to our lowest-pitched sample. Theoretically, you should be able set all loops made this way (cut-and-paste) to exactly the same length. But what will happen with higher notes is a greater degree of variance. So start out by using the longer loop length as a point of reference and gradually shorten the loop until the loop point is inaudible--shouldn't take long. A quick SDS dump to the Akai will wrap up this part of the sample editing process.

Before I move on to further editing the samples in the Akai, I want to point out that this process really does create a perfectly flawless sample. But the real trick is listening. Ever notice how EVERYTHING in life/nature/etc. works perfectly in theory but fails in practical application? The reason why our plans fall apart so often is that changing some of the factors involved and even unintentionally introducing outside factors causes variances in the result. For example, when sampling C4 and C6, I should have gotten identical results by setting the loop length to be the same as the C0 sample with absolutely no audible loop points. Playing back those samples did not yield the expected results.

So what do you do about it? Well, on the one hand, you can freak out and consult your visual editor to figure out how to exactly line up the end of one cycle with the beginning of another. I think a common misconception is that visual editing is faster than listening. Personally, I disagree. Scrolling back and forth comparing two ends of a wave is time consuming itself. A solution is using a loop tuner to compare both ends of a loop at once, scrolling through until you find a closer match. But upon listening, you'll find that the cycle is still audible. Visual editors are great for larger-scale sampling, but ultimately you'll want to be sure that what you hear is what you want everyone else to hear. A visual editor simply will not do the same job your ears will.

Now for my final observations! An SDS dump will quickly get small samples to the Akai. Don't forget that we're now dealing with much longer samples than the single cycles I was going for in the beginning, so SDS transfers are going to take a lot more time. At least the extra time involved is still reasonable, only a few seconds.

The samples still have to be edited on the Akai. The loops so far have been perfect in the few tests I've run. You'll want to rename them. You'll also want to set a root key. We're dealing with a small number of samples for each instrument we create, so many of the inconveniences we have to put up with aren't quite so bad.

Time to go make some more Thor patches. Perhaps my future blogs will be more about sampling with the Akai S2000! I've got big plans for the sampler.

"What's Love Got To Do With It?": Harmonica 1

The DX7 in all it's glory was, as most know, featured on a plethora of recordings back in the 80s. It's this sound designer's understanding that included THE harmonica sound on Tina Turner's "What's Love Got To Do With It?" If so, it's a real FM synthesis treasure.

As I worked on this, I was once again reminded of the vast difference between the DX7 and Thor. The Harmonica 1 sound, at least the way I envisioned it, is a bit richer than most patches I've written so far. And yet for it's complexity, it's really a simple patch.

We'll start with that. FM8 shows it as a 4-op patch. So judging between the visual data behind Harmonica and the actual sound, we see it's a DECEPTIVELY simple patch. No, there won't be any need for using a resonant filter as a 4th oscillator. But we do need all our modulation resources. I'm not necessarily going to attempt a direct emulation of the Harmonica structure. Rather, this time, I'm going for a somewhat imitative timbrel quality which will be modulated by three EGs and an LFO.

Let's start by getting the basic timbre down. We have a chain of 4 ops. We'll get a better approximation if we use Osc1 in place of A and B, Osc2 in place of C, and Osc3 for the carrier. Initially, we'll ignore using any FM amount in osc2 and 3. A and B have the ratio 7.62:5.056. What do we know about decimal values in Thor? Thor doesn't like them. So let's think about them in terms of notes in the C scale. 7=Bb, 5=E. 7.62 is equivalent or approximate to the leading-tone 7th, B natural. So we'll set the ratio for the nearest harmonic interval of a 5th--5:4, or G:C. To make this work for us, we set the semi to 4, raising them both a major 3rd to get closer to the ratio we need.

The next part arises from an ongoing complaint I have about Thor's implementation of FM. This specifically has to do with how Thor handles oscillators modulating each other and all the pitch instability it causes, radically shifting the pitch in most cases. This is obviously avoided by using Key Note to scale the effect. The inherent problem with this approach, however, is that the effect cuts out at C3 with scaling at 100. I'm not going to take the time to further fix the problem now for this patch, but I think what happens next is interesting.

Route Osc1, amount 100, to Osc2 FM. You'll need two scales. Set the first amount to 63, Filter Env, and the second to 60, Mod Env. Route Osc2, amount 68, to Osc3 FM, and scale first amount 40, Amp Env, and second amount 90, Key Note. What this does is link up all 3 oscillators without making them TOO perfect. I didn't use keyboard scaling with the first set because I thought a little instability and noise in that part of the chain wouldn't be such a bad thing, and I wanted to use the mod and filter envelopes to try to recreate a kind of breakpoint effect that the DX7 uses and is absent in Thor.

We don't need filters for this one. We do need a little vibrato, however, and this is where the designer's ear comes into play. The original DX7 patch has a dramatic vibrato whose rate is keyboard scaled (the higher you play, the faster the vibrato). I don't really like doing things that way, so I'm going to keep the vibrato constant. Set LFO1, amount 13, to Osc3 Pitch. Set LFO1 rate to 5.26Hz, triangle wave. Mod Env A 9.0 ms, D 3.2 ms, and R 3.2 ms. Filter Env A 78.8 ms, D 423 ms, S -3.7dB, R 7.50 s. Amp Env A 34.9 ms, D 4.73 s, S -13.4 dB, and R 423 ms.

To finish up, notice that FM8 oscillators are not perfectly tuned to harmonic ratios. The differences are mostly very slight and subtle. You can detune the oscillators to reflect that. For my taste, however, the timbrel variation strays to far from the sound I'm trying to get, so I just substitute the chorus effect. It should be very subtle, and I'll leave the reader his/her own judgment as to what it needs.

And that's it for a simple FM harmonica!

Complicated Relationships : Gong 1

So it's been a couple weeks since my last post when I was still fooling around with the Akai. That's going to be an ongoing thing, especially since I have much work to do. But it's time to get back to my original project: Thor versions of DX7 patches.

I have to remind myself of the importance of this project. It's not so much recreating DX7 sounds in the Thor synthesizer. It's more about what kinds of sounds we can spin off from them.

It's also about learning the synthesizer and coming up with ways of solving problems. We've already established that there can be no perfect replication of the DX7 architecture in Thor. We have, though, also established that by mimicking DX7 architecture (feedback loops, long chains or stacks of modulators and carriers) we can get results that approach the DX7 patch. I think that certain kinds of sounds are enhanced by these differences.

But sometimes these differences can make life really complicated for us. This week, we'll examine Gong 1.

Gong 2 is actually more difficult. But in looking at Gong 1, we can gather some clues that will hopefully make the variation a little easier to understand.

In making a synthesized FM gong sound, someone spent some good time working out how to emulate some of the nuances of a real gong. So imagine the sound of a lightly-struck gong. The initial sound is a pretty but deep, slightly ominous bell sound. Along those same lines, consider that the gong will have similar properties as bells and cymbals. This means a strong fundamental with unstable harmonics, maybe even some pitch instability as the gong settles into its core sound. There might also be a slow, gentle rise in those upper partials.

Translated into FM terms, this means at least two distinctively different envelopes--immediate attack and a slow attack, long decay times on both. We need a solid fundamental tone but with a first partial that is relatively close by. We'll have to be creative in picking out inharmonic partials and ratios and devise interesting ways of maintaining sidebands.

This is very much unlike any patch I've created up to this point!

Looking at the FM Matrix in FM8, we have a single carrier (at the bottom) modulated by 2 mod/carrier pairs and a single modulator. I don't normally consider this to be a problem. Big deal... Start at the top and work our way down, right?

Here's the problem with the usual routine: Look at ops A and B, C and D as mod/carrier pairs and consider their ratios. We have 1.2:1.4 and 3:0.745. OK, no problem. We'll simply use the chorus effect since this is obviously a product of detuning. But what about the last mod/carrier pair, the carrier being the only real carrier in this algorithm? Here the ratio is 0.8:0.5. This tells me that if what we're doing is a product of detuning, it's going to be a wild effect.

While I'll always champion the art of listening, I think what we need here is some basic mathematics to help us rebuild this FM gong timbre.

I'm not very good at math, so let's go ahead and get that straight right now! I'm not worried about getting everything exactly right: My satisfaction is in whether or not I like the results. We have to start with the core sound, and since the other oscillators will be routed to this one, let's start with oscillator 3.

Here's what I did: To get something like a ratio of decimals, I went ahead and multiplied the numbers by 10 to get 8:5. The 5 (carrier ratio) represents a tone that is 1/2 the fundamental--in other words, an octave lower. Here, however, I like to think in terms of a pitch class relative to scale, quite simply, in the key of C. In this case, it would be the note E. 8, by contrast, represents the note C three octaves up. What we're really talking about here is the difference of a minor 6th. So what we need to do is set the octave not 3 but 4 octaves down (oct 0) and semi up by 8 (to account for the minor sixth). That will convert the carrier ratio back to 0.5.

As you can already see, a top-down approach wouldn't take this kind of tuning into account. As we've succeeded in created a core sound by looking at the math, we'll need to continue on in the same way to get the proper ratio for oscillator 2.

The ratio here is 3:0.745. Note the carrier ratio. Thor doesn't allow anything less than whole numbers, but in some cases we can use approximations. Let's round this up to 0.75. What do we know about this? Well, 0.5 is an octave lower than fundamental. That means 0.75 (halfway between 0.5 and 1.0) is analogous to a perfect 4th below fundamental, or the note G in the key of C. What about the 3 mod ratio? Simple: It's a 12th up, or the note G. This is serendipitous. All we need to do is tune these operators in octaves and reset the oct and semi settings to get a note a 5th above our lowest partial (octave below fundamental). Very simple. Set carrier at 1, mod at 4, oct 3, semi 7.

Now let's look at the top of the chain. The ratio is 1.2:1.4. This one took me a while. I decided to look at the decimals as 10ths of an octave. This is unusual in normal circumstances because we tend to think in terms of octaves divided into 12 equal parts. We do have one convenient way into re-dividing the octave, though. The distance from one semitone to another is divided into 100 cents. So 1 octave=12 semitones. 1 semitone=100 cents. Add them up, and you get 1 octave=1,200 cents. That means that 0.1 octave=120 cents. Makes sense in theory, but how do we put this in practice?

Let's look at that ratio again. 0.2 octave (do the math)=240 cents=2 semitones+40 cents. 0.4 octave=480 cents=4 semitones+80 cents or, in Thor terms, 5 semitones-20 cents. What we're looking at here is sharp D and a seriously flat F, so somewhere in the neighborhood of a minor 3rd. This could easily be done now that we have the information we need. So even though we won't get exact results, I think what we'll have will be interesting.

To do this without resorting to some severe octave tuning not possible in Thor, I went with 6:7, or G:Bb. To get it in the right octave, the setting has to go one more lower than our target. That means 3 octaves lower or oct 1. Semi has to be a perfect 5th up, so semi 7. I chose to further tune this just 20 cents sharp.

I suppose I could also have gone 8:7 and brought the interval even closer together. It's just a personal preference of mine that inharmonic ratios have a little bit of spread to them. When I say inharmonic, remember that Thor doesn't do inharmonic ratios. They have to be created artificially within Thor by using combinations of FM oscillators that are detuned relative to each other enough to count. So even though the ratios of one oscillator are harmonic ratios, they are heard as inharmonic when used in concert with other oscillators and displaced by an octave or other interval.

Now that we are beyond our relationship issues, we need to tie everything together. Keep in mind when you use an osc to mod another osc, you should use two scalings: An envelope and a voice key note. Scaling by voice key helps alleviate some of the pitch instability that occurs in this line of work. I'd also normally route FM pairs to osc 3 pitch, but after experimenting with this for a while, I conceded that the gong sideband effect just wouldn't be pronounced enough where I needed it in the spectrum. So here's the routing: Osc2, amount 37, Osc3FM, scale 100 Mod Env, 100 Key Note. Osc1, amount 30, Osc3 FM, scale 100 Mod Env, 100 Key Note. Over in the "single" section, Amp Env, amount 29 Osc3 FM Amt; Mod Env, amount 57, Osc2 FM Amt; Amp Env, amount 28, Osc1 FM Amt.

The Mod Env. settings are: Delay 1.12 sec, A 4.63 sec, D 11.9 sec, R 10.8 sec. Amp Env. settings are: A 1.9 ms, D 19.2 sec, R 552 ms.

Only Oscillator 3 is routed to Filter 1 (bypass). You might try applying chorus to this one, but I preferred using the Delay only with no mod amount. The lower registers are more effective and noisy (don't stray below C2) while the higher octaves have a very bright, pretty asian bell kind of character. The synth crescendo is less noisy here, but doesn't really sound much like a gong. As always, these notes could be used for other kinds of effects or as inspiration for more variations of this sound.

Back to work: Sampling with the Akai S2000

As soon as a create a schedule, I'm doomed to fall behind. Maybe that's for the best!

Anyway, I got some good get-to-know-you time with my Akai S2000 yesterday. Considering what I want to do, my goals in using a sampler, I've made some observations which I'll share here.

I decided for the sake of practice I'd create an additive waveform in Absynth and sample that directly to the Akai and do all the necessary edits on screen. Now, if you aren't familiar with the Akai S2000, you need to know that the only user interface is a tiny, 2-line LCD with that classic green glow. There is no graphic editing whatsoever, and this came as a complete shock after having used NN-XT for so many months now. Actually, the Casiotone side-project has been the first in-depth use of NN-XT for me. It's easy to set the zones, automap, and so on in NN-XT. Using the trackpad or my handy trackball mouse, fine-tuning is easy--on the head, on the eyes, and on the wrist!

Not so on the Akai. Anyone out there who takes the plunge and gets an Akai S2000, here's a little advice: My new eBay friend who sold it to me sent a disk with MESA, which is a graphic editor for the Akai S-series samplers. He told me I should pick up a Mac with SCSI running OS 9.22. Absolutely right! If you go the route of buying an old sampler like the S2000, you'll probably want a graphic editor. So go ahead and bid on an old Mac while you're at it.

I'm too poor, though. I'm going to stick with original plan and program this thing right from the face of the machine.

So I figured out how to record a sample. I'm not to the mapping phase yet, just doodling around. I also figured out how to loop that sample. The steps to do this aren't difficult. I sampled an additive sound from Absynth to start with. Here's the hard part: Akai has a "find" function that will find loop points for you, except they aren't really that accurate, at least not the way I'm working. I have to set the "loop at" point all the way at the end of the sample. Also, I found this works best (for me) if the entire length of a sample is 100 ms. The loop length is set at 0.

Here's where I unhappily discovered one of the most useless features of this sampler. The loop length apparently is in samples, but there are 3 decimal places after that. I'd like to know why! I adjusted that thing with the data wheel until I thought my thumb would fall off! Useless. So I went back to spinning around through samples until I found a proper loop point. Because I've practiced this in NN-XT without a graphic editor, it wasn't long before I found a perfect fit.

On to my next experiment: I created another wave in Absynth, loaded it into another oscillator, and detuned it for an inharmonic effect. I sampled it. Before I continue, let me just say that I've never been diagnosed an Aspie, but I think it's obvious that I am one. While I do enjoy the effects of my creative focus, some of the more harmful effects of the condition came out during this experiment. I must have spent hours trying to find a loop point! Now, if there are any more experienced sound designers out there reading this, you're probably laughing because you already know what I found out yesterday: inharmonic sines will never line up, which makes it impossible to find loop points in a few cycles. Mathematically I'm sure it can be done, but I'm no good with math.

Solution: Sample each Absynth oscillator separately and create a layered sample instrument in the same way as the Absynth patch. Even though this eats up just a little more memory, it's worth it. This will create some lovely bells and similar pad sounds. I also found that by re-detuning the second wave you can create different inharmonic effects. I'm talking here about complex waveforms, not generic chorusing that we normally do by detuning. Take this and run it through Mainstage to add chorusing or other effects, and you got quite a powerful little instrument!

I'm still fumbling around right now and I still have so much to learn. Time to put on a pot of coffee and get to work...