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.
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