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About this blog

This is essentially where I store bits and pieces of things I'm learning or revising -- primarily sound design, production, and music theory. I was keeping these in a private locker at home, but decided to start sharing with the Muse once Alistair and PaulC turned me on to the blogs section. You may find it interesting/helpful, you may not. But it's all in good spirit. 

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Okie dokie.... I am not feeling well today, but I tried to dig into this Retro synth so that I could learn what wavetable synthesis is. I did not have a lot of fun, which at first I blamed on wavetable synthesis 'cause, you know, what a bastard, but later when I started to understand it better and actually have a little fun, I recognized that I just feel crappy altogether. I shouldn't have gotten the fish. Didn't Striker teach me anything when I was a kid? ("A hospital? What is it?") Ghetto fish. Pulled a knife on my 胃 and there's about to be a bacterial showdown. Maybe. Oh, sorry. Peut-être. French group tomorrow. I better get in character. ハク〜〜〜


First things first. I consider the Retro synth to be a journeyman synth on our little Let's-Finally-Get-to-Know-Synths! tour. Most of what it involves are things that we have already learned through the previous synths. Although, in this synth we are using graphical representations of envelopes instead of ADSR sliders, so that's cool. There are a few other minor new bells and whistles, but I'm not going to get into them as they're pretty obvious all in all when you use it. 


TBH, this synth is not really selling me. It took a lot of poking and prodding to find it interesting. But it has one big perk: It's a 16-voice, two oscillator synth, containing the following 4 engines: 


1) analog - classic sounds (leads, pads, basses)

2) sync - aggressive sounds (leads and basses)

3) wavetable - variety of synth and simulated natural sounds

4) FM - recommended for bells, electric piano, clavinet, and spiky bass sounds


We've gone over "analog" subtractive synths, and we've done an FM. There are two new ones here. 


What makes the sync engine interesting is its singular "Sync" knob. This knob changes the starting point of both oscillators. Fini. 


Now the wavetable engine... that's what we're here for. So wavetable synthesis takes an audio clip and chops it up into, in this case, 100 "tables" in a sort of circle. Each table in my understanding is a wave cycle or period sliced from that audio sample. Now these are obviously too small in respect to your original sample, so you're not going to get anything remotely like what you put in. Instead, these cycle slices are going to be used in lieu of the general sine/triangle/sawtooth/square/etc. waves that we've been using in the other synths. When you load up a wavetable, it puts in on both oscillator controls, here called Shape. Moving through the oscillator controls chooses different tables, one at a time. So, moving through these you can try to find a new and interesting waveform to generate your core sound with or mix two different waves. 


And that's pretty neat, I guess. There were some cool sounds here and there from it. But where it gets really interesting is when you use that shape modulation knob to route either the LFO or the envelope into the shape "oscillators". What happens then is that the tables/cycles being played will change over time, giving interesting movements. So I had a little fun with that for a while. I even tried creating my own tables from audio files. There's plenty of room to experiment. Anyway, like I said, I'm not feeling well, so let's wrap this up. Note that you can slow down the movement through tables, so as to not get such an exaggerated effect as these. 


Examples of using the envelope to morph through the tables being chosen for oscillation: 

Wavetable Sample 1

Wavetable Sample 2

Wavetable Sample 3


In this last one, I just hung on a couple notes while I manually shifted one of the shape knobs, getting it to choose different tables/cycles as an experiment: 

Wavetable Sample 4


And here's the synth. It's on the table engine mode, so you can't see Analog, Sync, or FM. Suffice it to say, with the exception of FM, their controls are not so different. Of course, the fundamentals of what they're doing are quite different, however. 





ES1 Synth

I'm just pasting my notes in for this one. I'll organize them at a much later date. 
This synth introduces several new concepts from the others, but there are four big things I see:

1) LFOs and other envelopes that can modify a great variety of parameters beyond gate and volume, including modify each other.

2) Low and high arrow specific, dial-in definition sliders for several parameters that work in conjunction with velocity and the mod wheel.

3) The ability to take external signals via side chain and route them to be the sub oscillator or LFO modulator.

4) An internal feedback system for the resonance filter. 


Cranking up the resonance for feedback gives tones that are tuned by the cutoff. For fun, I randomized the LFO and routed it to the cutoff, killing the main oscillator, and got sounds just like you might hear on cheesy old SciFi shows. Getting slightly more musical, I created this little instrument: 


Digital Bubble Tones


Now, the side chain input... I routed some waves in it, and also a 12-string guitar, and a violin... nothing was all that interesting, to be honest. I mean, you can put like a drum loop through that sub oscillator and play around with the filter if you want. That wouldn't be unusual. Now most work of this kind, if you're privy to it, you can just do with a regular filter. But there were a few things kind of interesting. Here's me messing around with parameters while a drum loop is side-chained in as that oscillator wave: 


ES1 Side-Chain Percussion Test


What I find interesting about the above side-chain experiment is that you can play the main oscillator with the side-chain feed, and whatever sound changes you make to the oscillator will be copied by the side-chain audio. So... I can kinda see having a little SD fun with this one. It should be noted that I had the source track muted, and the feed from the side chain only plays when you hit a MIDI note. (In the example, you can hear me just moving between two notes, I think a 6th apart.)



Anyway, here are my notes. (And remember to put a limiter on the track so as not to damage your monitors or your ears. Double it with one on the Master if you want to be extra careful. While I was jumping through modulation routings, more than once there were sudden nasty sound spikes!):


The ES1 is a subtractive synth emulating analog circuitry. There are two oscillators, one of which is a sub


Note that it has an analog% option in the global parameters on the bottom left. This is actually to simulate the wear of analog circuitry. A higher percentage equals slight randomness in note and cutoff frequency. The manual suggests 0% for more percussive sounds, and higher freqs warmer analog sounds. 

You can change the waveform of both oscillators. It can be interesting to mess with the sub. It also has the options of generating white noise, or routing a side chain signal through

The sub square/rectangular(?) can be one or two octaves below. The pulse wave is two. (Backwards?)


5 pipe organ pitch settings

Change square wave width (actually moving between different pulse waves)


Can have different pitch ranges on the bend wheel, using a combo of global and extended settings. (Neg. Bender Range)


Legato engages portamento. Portamento speed is set by Glide. 


Difference between a high voice setting and “full” is that, if you have a high Release setting in the envelope, chords and notes will not overlap as they are played and released at “full”. This automatic release cutoff can prevent messiness sometimes. 


Like the ES E, there are three chorus settings. 


The ADSR setting below gave a type of “reverse effect” feel to the notes played. It is useful to get to know envelopes better, especially as they are used for MANY instruments and effects, including those non-synth. 


The ES1 can self-oscillate by driving the resonance to maximum. This creates a sine wave, adjusted by cutoff, envelope, and various settings. Note that it is not dependent on musical notes pressed. To hear it in action solo, turn the sub oscillator off, but the mix all the way to sub, and put the resonance at full. (It will actually kick in around 75%)

Try getting feedback as above, and then putting the LFO on the random wave, high speed, and cutoff. You will get old school SciFi bleeps and bloops. Remember that the cutoff is what really changes the feedback pitch. Experiment with intensity widths on the LFO effect. 

While we’re at it, with these settings, engage the sub oscillator by changing to an actual wave. You’ll get interesting effects from the LFO randomizer. 


The big “Filter” word in the middle is actually a draggable zone, to mod cutoff and resonance at the same time, somewhat like an XY pad. 


This time, the low pass filter has four types (would be beneficial to have a quick discuss and show on low-pass EQ): 

24 dB classic — mimics a Moog filter. Increasing the resonance reduce low end

24 dB fat — like an Oberheim filter, it tries to compensate for reduction of lows when resonance increases

12 dB — a soft, smooth sound like the early Oberheim SEM synth

18 dB — similar to the filter of the Roland TB-303


The drive slider is input level into the filter, changing the resonance response. (Notice that in the extended parameters is “Filter Boost”. This increases the output of the filter by 10 dB, but at the same time lowers the input/drive by 10 dB. Ultimately, this gives different sonic characteristics to the filter, especially at higher resonance values, greatly increasing feedback. 

The key slider is like the 1/3, 2/3, 3/3 buttons from the ES E(?). On zero, all the keys on a keyboard react the same. The more you move towards zero, the more the sound follow pitch/note, similar to a real acoustic instrument. Specifically, there is a relationship between cutoff frequency and pitch. This difference is obvious when you play a single note at zero and then again at max. 


Parameters with upper and lower arrows

You are setting a range with the two arrows. This is useful for something that gives variable input. Most of the double-arrow sliders here are for velocity, your MIDI keyboard (or whatever instrument) note “level”. So, whatever you set the lower arrow to is the range correlating to velocity 1 (as soft as you can play). Similarly, whatever you set the upper arrow to is the range correlating to velocity 127 (as hard as you can play). The ranges in-between will match your playing styles between 1 and 127. Dragging the blue bar between the arrows moves both at the same time. 

Note that the lower-left slider on the LFO listens to your mod wheel on your keyboard instead of velocity. 


There are three buttons to select what ADSR parameters you want to affect the amplitude


Something that should have been mentioned before: Attack and Release times of zero can create pops in when you play, so it’s generally best to have them up a little bit, even if you want an immediate attack or release. 


Note that when playing in legato mode, whatever velocity you trigger the gate is where the gate will stay while you play in legato, no matter the velocity you hit later keys with. I think I’ve said this before in different words. 


The manual suggests setting the cutoff low and the resonance high, then move the ADSR via Velocity arrows around to get to know it. 


The FM modular routing setting creates FM-style metallic sounds on feedback. Otherwise, it’s a distorted effect. 


The LFO introduces a descending sawtooth

The “random” wave is actually called a “sample and hold” wave. The one next to it is also a random wave, but smoother. 


Setting LFO rate to zero = DC, which means the mod wheel is to control the change. However, I have yet to find this very pronounced. There are some effects, like if you have the main osc on square wave and have the LFO routed to pulse width, rate zero, at high mod wheel zones you can heard the change in pulse width. 


The Mod envelope form; “full” is static/off, decay is fade-out, and attack is fade-in. Small settings can be percussive. Otherwise this is set via velocity parameter definitions. 


From the manual: 

Set up a delayed vibrato

  • Drag the Form slider to the right—toward attack.
  • Select Pitch as the LFO target in the left column of the router. 
  • Use the Wave knob to select the triangular wave as the LFO waveform.
  • Drag the Rate field to an LFO rate of about 5 Hz.
  • Drag the upper Int via Wheel arrow to a low value, and the lower arrow to 0. 


Here's a slightly cropped pic of the synth:



(This was something I typed up years ago when I first began writing music. I did it both as a "note to self" as also as JFL practice.)
TL;DR -- Nature is super cool but freaky weird. Oh, and don’t overlap too many rich 3rds as you might accidentally generate some off-tune ghost notes that’ll kill your compositional buzz.
General advice when orchestrating or arranging is that you should be careful about layering 3rds. By 3rds I mean for example in a general 1 3 5 major chord sequence, the 3. So in A major (A C# E) this is the C#. If you layer these you can sometimes create undesireable wolf notes or ghost notes - notes that can be heard, but are not being played by any instrument.
一般的にオーケストレーションや編曲を行う際に気を付けなければいけないことは「第3音」の重ね方です。「第3音」というのは一般的な1-3-5のコードの「3」の音のことです。例えばAメジャー(A C# E)では「第3音」は「C#」です。この「第3音」を重ねてしまった場合、どの楽器からも演奏されてない音が聞こえてしまう「ウルフノート」とか「ゴーストノート」と呼ばれる好ましくない音が出てしまうことがあります。
The truth is, these notes are always there, but generally blend into the rest of the sound spectrum and are not heard individually. But sometimes, they conflict with dominant notes that are being played, and this is where you have a problem.
I ran into this when I was mixing a passage that hung on an A dominant chord (A7; A C# E G) for one measure. I had a string section, an organ, and an acoustic guitar playing. Every time it came to that A7 chord, I heard this horribly grating G# sitting right in between a G and an A. I muted all the tracks and started going through each instrument one at a time to find the culprit, but none of the instruments were playing a G#. Despite this, when I put them all together, I could hear the G# clearly.
私も弦楽器、オルガン、アコースティックギターで構成された曲の1小節の中のAドミナントコード(A7; A C# E G)のミキシングに苦労していたときにこの問題に遭遇したことがあります。このA7のコードの部分に差し掛かるたびに、ひどくきしむようなG#の音がGとAの間で聞こえるのです。そこで、すべてのトラックをミュートしてすべての楽器を一つずつ聞いて犯人捜しをしましたが、どの楽器もG#の音を出していませんでした。それにもかかわらず、すべての楽器を一緒にプレイするとG#の音がはっきり聞こえたのです。
Why? It's because of the physics of sound, and what we call fundamentals and harmonics.
You see, technically when we play a note on an instrument or sing it aloud, that single note contains all notes. (The universe in a grain of sand, eh?) When I pick up my acoustic and hit that E, I hear the E, but buried underneath it is a B, and another E, and more Bs and G#s and deep within D#s and deeper and deeper C#s and As - and Cs and Eb even if you wanna get down to it.
And these as you may know are the overtones or harmonics of the note. The picture to the right (taken from Wikipedia) illustrates a fundamental (low C) and its first 7 harmonics. (Note that a lot of classical composers like to use these chords, especially on a big fat cadence.) All instruments produce overtones, and it's the different quality of these overtones that creates the individual timbre for each instrument. (I suddenly am picturing the vast field of humankind.) The first harmonic is generally the octave, then going higher is a fifth (or 12th, as it is an octave higher), then the second octave of your root, then a high third, another fifth one octave higher… and so on into an infinity that our ears and minds don't pick up on in a manner that separates them individually.


これらは倍音といわれるものです。右の写真(Wikipedia から引用)は基音(ローC)とその最初の7つの倍音を示しています。(注.多くのクラッシック音楽の作曲家はこれらのコードを好んで使います。特に重厚なカデンツなどに用いられます。)全ての楽器は倍音を出し、質が異なるこれらの倍音がそれぞれの楽器の異なる音色を生み出しているのです。(世の中に色んな人がいるように。。。)一般的に最初の倍音は1オクターブ上の音で次は第5音(または第12音・・第5音の1オクターブ上だから)そして基音の2オクターブ上、2つ目の「第3音」、1オクターブ上の「第5音」・・・と私たちの耳や脳がそれぞれの音を聞き取れないところまで無限に続きます。
Try taking an acoustic instrument like a guitar or cello for example, and hitting one open string. Sit and listen to the sound of the string change and develop. Listen for other notes. Try other strings - some have stronger harmonics than others. What can you hear?
On that acoustic of mine, I hear the 5th on the A string pretty clearly after a few seconds. And this makes sense being that the 5th/12th is the strongest harmonic after the first octave.
So let's go back to that troubling A7 chord we were talking about at the beginning. On going through the all the instruments playing the A7, I realized that there were a lot of 3rds being played. The 3rd of the A7 is a C#. Remembering what we just learned about harmonics, we know that the 5th is often strong. Well, the 5th of a C# is G#. And whenever you double any note, both the fundamental and the harmonics get stronger and louder. There was the culprit. In particular, I had the organ playing extra 3rds (C#). Organs are harmonically rich instruments, which means that their harmonics are more pronounced than those from other instruments. (While we're at it, distorted instruments, like rock guitars, are also harmonically rich.)
では話を元に戻して私が体験したA7のコードの問題について考えてみましょう。 その後A7のコードを弾いている全ての楽器の音をチェックしたら、たくさん「第3音」が出ていることに気づきました。A7の「第3音」はC#です。これまで倍音について勉強したことを思い返すと第5音が強いことはわかりますね?C#の第5音はG#です。どのような場合でも音を倍にすると、基音と倍音は強く、大きくなります。ここに犯人がいたのです!特にオルガンの演奏は余分な第3音(C#)を出していました。オルガンは倍音が豊富にでる楽器で、それは他の楽器と比べると強くでます。(参考としてロックのエレキギターなどの歪んだ音色の楽器は倍音が豊富に出ます。)
The solution was to remove extra C#s. I did that, and the G# disappeared.

EFM1 Synth

So now we move out of subtractive synthesis and into Frequency Modulation synthesis with the EFM1. (There is no existing EFM2 that I know of. Must be a legacy thing. Unless the 1 is because there is only one modulator operator.)


Little historical note: Apparently FM synthesis, spurred by John Chowning's 70s paper, The Synthesis of Complex Audio Spectra by Means of Frequency Modulation, the patent of which was picked up by Yamaha, marked synths moving from analog to digital. Euh, whaddya know... 


The most noticeable difference with this mini-FM synth and the previous subtractor synths is the ability to generate exaggerated harmonics on the oscillators/waves. (It should be noted that FM oscillators are called operators. Oh… light bulb moment on Ableton’s Operator instrument…) 


So the basis of FM synthesis is that you have one waveform modulating/modifying the other. I have to admit when I first heard this description I was unimpressed. I mean, in the other synths for example, when we play more than one wave at the same time, they’re modifying each other. That’s how sound works. So I knew there had to be something else going on. 


Let’s take a commercial break so I can dig a little more into that…. 




Introducing:  Cold War UNICORNS!! 


Pasted Graphic.jpg


It’s a Cold War showdown! This play set allows you to play out the intense struggle between two global superpowers in the majestic fantasy world of the Unicorn! Can the Commie Unicorn’s horn of classless social structure hold up against the Freedom Unicorn’s hooves of capitalist opportunity? You decide! Each hard vinyl figure stands 3 3/4-inches tall and features articulated joints for all sorts of dramatic poses. You’ll love them, they’re educational and absurd! Like the real deal! 



/commercial break


Okay, so while in the subtractive synths we were looking at before different waveforms were mixed together, in this FM synthesizer, you’re only outputting one waveform (the carrier), but that is being directly modified by another waveform (the modulator). It’s worth noting that while the EFM1 only has one modulator, more advanced FM synths will have multiple modulators effecting each other often in a chain before hitting the carrier for the final sound output. 


Aha! The modulator harmonic setting functions in order just like real, natural harmonics. I noticed that when I moved it up a notch, it actually jumped an octave. The next notch brought a fifth, then a fourth (two octaves from source), then a third…. It must move all the way up through natural “Pythagorean” harmonics. This was noted when the carrier was on 0, not affecting the ratio. 


(Note to self: write an article about fundamentals and harmonics.

(Note to note to self: Screw that, just copy over the J<>E one you wrote back in Japan.


What’s happening is that the modulator and carrier together are creating different tuning ratios. (See the bottom of the BreakTweaker post.) The harmonic settings create different ratios, and the FM intensity mods the amplitude of these ratios. (So does the mod envelope sustain.) The carrier is a sine wave by default, and the modulator is “multiwave”. Note that very small changes to these waveforms can generate wild results, unlike the subtractor synths. 


(Note to guitarists: Ring modulators are basically the stomp box “organic” version of FM synths, also working with a modulator (the guitar) and a carrier-sine signal.)


We’re going to go straight from the manual for a bit: 


“The carrier frequency is determined by the played key, and the modulator frequency is typically a multiple of the carrier frequency.


You can tune the modulator and carrier to any of the first 32 harmonics. The tuning relationship, or ratio, between the two significantly changes the base sound of the EFM1, and is best set by ear.


You use the Harmonic knobs to set the tuning ratio between the modulator (left) and carrier (right) oscillators.


In general, even tuning ratios between the carrier and modulator tend to sound more harmonic or musical, whereas odd ratios produce more inharmonic overtones—which are great for bell and metallic sounds.


In this respect, you can view the tuning ratio as being somewhat like the waveform selector of an analog synthesizer.


Experiment with basic tuning ratios

Do one of the following:


  • Set the modulator and carrier to the first harmonic—a 1:1 ratio.
    A sawtooth-like sound is produced.
  • Set the modulator to the second harmonic and the carrier to the first harmonic—a 2:1 ratio.
    A tone that sounds similar to a square wave is produced.”


Muchos arigatos, Mr. Logic. 


To make this simple little FM synth more complex, you can choose a variety of different wave forms as the modulator operator. While the full left setting is a classic sine wave, as you turn it up, the waveform changes smoothly. It actually sounds similar to tuning an old radio, which makes sense, cause I mean, you know, frequencies. Anyway, some very interesting and unexpected sounds can be generated here. 


I think that’s really what I find different about this FM synth over the subtractive synths—like I mentioned before, small tweaking of the settings can yield wildly different sounds. There’s a certain unpredictability to it all, and it definitely will require more tweaking to get to know it. I don’t imagine that I can try to reverse engineer sounds with this one until I spend much more time with it than the others. I think for starters I’ll just look at the presets and see what sort of ratios and modulator wave settings they are using. 


Oooh yeeeah… just with a ratio of 1:1, I’m getting some nasty-cool bass sounds out of this thing. (I need to map this to a MIDI knob. It sounds really cool to change the modulator waveform while you’re playing. Oh, and then turning up that big FM intensity knob in the center and continuing to tweak the modulator wave makes some really phat sounds. It’s beginning to sound a little dub step a la that Skrillex ボコボコ.) I should mention, btw, that there is a (one 8ve down) sub-oscillator built in with a volume knob. While low ratios of the synth by default create some thick basses, as you move up the harmonic spectrum, it quickly creates high pitches. Turning up the sub-oscillator can help you recover lost low end. But, well, deep low end. 


Getting some cheesy metal guitar synth sounds out of this thing messing with the waveform and intensity at a 1:1. I feel like I’m playing an old Nintendo action game like Metal Gear or something. 


The “Fixed Carrier” button is supposed to uncouple the carrier wave from being modified by anything including the keyboard, but I’m still getting pitch variation. That must be coming from the modulator operator, which is still affected by the keyboard and LFO, etc. 


The “modulator pitch” knob adds a pitch effect to the envelope. All this really teaches us is that, like the previous envelopes that could control both the filter gate and overall volume/amplitude, any envelope can have multiple sound parameters routed through it


Note that legato modes are specifically set up to not retrigger the envelope when playing with legato. Of course, I think we know this by now, but I just wanted to print it. 


Other notes: a Unison button and Stereo Detune knob can be useful to create chorus-y sounds. And like the other synths, there is a velocity sensitivity knob. I like this, personally. It allows your playing style to influence how heavily the envelopes are used. 


Finally, and this is kinda neat, there is a randomize button on this synth, complete with a percent control of how wildly it will randomize. Hitting this a few times yielded completely different results, including one that sounded like a bird. 


Okay, so… while I’m still not a big synth fan, I am more and more seeing both their usefulness and coming to understand better how to build sounds with them. We’ll get around to more of this later. Same Bat-channel. 


Screen Shot 2018-01-17 at 5.58.02 PM.png


ES E Synth

The third synth we're looking at in LPX's legacy simple synths is the ES E(nsemble).

The ES E is an 8-voice subtractive synth. It's probably the step up from the ES M, as opposed to the ES P. 


Notes about this synth: 

1) The waveform generator has three settings: sawtooth, square, square with greater widths as you turn it up. So, basically this is good to hear the changes that happen as you increase the width on a square wave. 

2) If you increase the width of the square wave while you're playing, it gives a mild bit-crusher effect reminiscent of old video games. (That probably did exactly this.) 

...actually, when you think about it, bit-crusher effects are essentially doing just that. When you lower the sample rate, you are sort of creating greater "width of squares" in the computer-viewed sampling of your audio. 

(Note to self: write blog post about sampling rate vs. bit depth. Though I'm sure you guy have it down already. If not, this article is decent.)

3) The LFO (which it doesn't seem can be completely turned off), can be set to modulate the width of the square wave. 

4) Resonance seems to have no effect if the AR gate is not open. 

5) There are three simple chorus effects that can be turned on/off. 


Playing with the width of the square wave, and also with the LFO modulating that width, is interesting to view in a frequency analyzer. While a small width setting shows a waveform with a lot of frequency "hills", as you turn it higher, those hills themselves widen and become fewer, until finally at the highest setting there is basically only one hill. 


All in all I gotta say I don't find this synth very interesting. Actually, the sounds it creates are exactly the cheesy old synth sounds I tend to not care for. (Although I will be quick to admit there are other artists out there who can make some really cool tunes using these "cheesy old" synths.) 






ES P Synth

Annnd next on the list is the ES P, which stands for... wait for it..... Emagic Synthesizer Polyphonic. Whoa... Bet ya never saw that one coming. :P


Like the ES M, is it subtractive, so essentially you get the synth to generate waveforms and then cut/subtract frequencies from them with a low pass filter. An ADSR gate time/movement modifies how the selected frequencies are cut. 


There's one thing I loved when messing around with this guy. With the frequency view up, I could see and hear so clearly how each knob affected everything. I could both see and hear the differences between the triangle, saw, and square waves. I could watch the frequency viz transitions as I played different notes. I could see how wah modded the resonance and vice versa. And best of all, playing with the ADSR envelope was like writing a story. It was very clear in sight and sound how the controls affected the sound over time. Very useful training tool! 


So attack and decay on this one control how quickly the sound sweeps up through the frequency range (the ADSR envelope is essentially a gate on the subtractive filter), starting from the frequency set. (It also seems to affect amplitude.) The range seems to be set, which means if your frequency is low, it will only go so far, but if your frequency is high, it will go "all the way" and maybe hang there briefly before coming back down. How long depends on your attack and decay settings. Sustain controls what frequency level it hangs at after the decay. There is a direct relationship between sustain and the frequency knob, meaning that so far it seems that I can achieve with one what I can with the other, with the exception being that the sustain will also affect the sustained volume, not just the sustained pitch. 


Release is simply depress decay. 


Alright, some interesting things about the ES P: 

1) It has three pipe octave settings like the ES M -- 4, 8, and 16. 

2) Along with the triangle, saw, and square waves, it has  -1 octave and -2 octave square waves. All waves operate on sliders so you can freely mix their sounds. There is also a noise slider

3) An optional LFO can either affect wave vibrato or the frequency pitch a la wah wah. 

4) There are 1/3, 2/3, and 3/3 keyboard follow buttons, which tell the ES P filter to follow the pitch you hit, keeping a constant relationship between the frequency and key. 3/3 turns it on for the whole keyboard range. 


There's also an knob for distortion and one for a stereo-spread/chorus effect. 


And that's pretty much it. Of course, it may not seem like much is going on, but already, there is plenty of room to create a wide variety of sounds. 


Party on, Wayne. Party on, Garth. 


PS~ Another interesting thing about these little synths is that the knobs are off in their central position. Moving them left or right has different effects. The ADSR intensity knob, for example -- moving it to the left reveres the direction of the gate, so that it cuts as it moves instead of sending a wave that opens up frequencies. 


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ES M Synth

I did not like the 80s. But, I mean, those were my early formative years, a time when my gut had the emotional content of liquid nitrogen. Or was it mercury? The crow's in the cupboard, Jimmy! 


Perhaps that's why I never got into 80s music, despite that it really seems to be a favorite of many people. It has that strong nostalgia for others that to me was always more 不味い。And the 80s was a time of the big stab. Not you, Karasu-san. I mean those cheap synth sounds that dominated. 


So... I am behind on learning synths. I'm more of a play-around-with-samples kind of guy when it comes to software tools. But there are some fantastic synth sounds out there, not to mention great tricks with synthesizers. The deep whisper of a throbbing, ghostly synth bass under steadily churning orchestral waters? Yes, please. 


I am astounded by the amount of content that comes with LPX. And it just keeps growing with all these free updates! It recently came to my attention that LPX has nearly a dozen synths built in, and several of them are really bare bones. Might be a good place to start learning a synthy thing or two. Those full-fledged ones can be a bit discombobulating for me ol' tête bête.


Let's start with the ES M.... which I believe stands for Emagic Synth Monophonic. (If you guys remember, Logic was created by Emagic.) It is a subtractive synth, which is supposed to mean that you are fundamentally cutting away frequencies from the waves to create your sounds. Right? 


The ES M has only a saw wave and a square wave, with the square tuned an octave lower. You can blend between the two. There's also three little buttons on the left that are neat mainly because of what they signify. You must choose between 8, 16, and 32. These stand for pipe organ lengths. The longer the pipe, the deeper the sound. (Ever played a pan flute?) 


There is a very simple filter with low-freq cutoff and resonance. (Must be the subtractive part.) There are very simple decay and volume controls that can be affected by "intensity" knobs and velocity sensitivities. And there is a distortion knob. Fini. Pretty simple, right? 


Oh, I should also mention that it has portamento always on, and there is no way to turn this off that I found. So if you play legato, you'll get slides between the notes. You may want this, you may not. There is a glide setting to control how fast the slides occur. 


So, I just loaded up a track with this little synth, and created a looping line that had some variation in pitch, melody, and dynamics. The reason being that I wanted to just tweak the synthesizer and listen, without having to be concerned with playing lines. Also, I opened up a couple of frequency visualizers to help put things in my mind's eye. (Us upright walking, eyes in the front humans are largely visual learners. And even if you aren't part of the majority, meaning whether or not you're a visual learner, having multiple avenues for your mind to absorb information can be beneficially compounding.) 


So, this was a cool little synth that seems to excel at bass stuff. You can make simple monophonic lead lines with it, but I don't know, I didn't care too much for those sounds coming out of it. It's like Michael J. Fox calling me to warn me that Leonard part XVI is going to be written from a funkadelic neon-Pluto Hollywood jailhouse. 


Btw, as you can see in the pic below, you can have the LPX manual open and floating while you work in LPX. Very useful for learning. 

EDIT: Oooh... you know what would probably be a good exercise? Duplicate the tracks, then go through factory settings on one track, while you try to build / reverse engineer it on the other track, A/Bing the two. 

EDIT part deux: Yeah, I can confirm that A/B reverse engineering the synth sounds is a good way to learn. Mess around with it and make your own sounds to get familiar with the controls, then try A/B building. With this synth it's pretty simple, but it's still a good way to befriend nuances. Although to clarify: the whole frequency visualizer thing to learn is only useful when tweaking some parameters. When it comes to rebuilding the synth presets, use just your ears. 





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I'm in a variety of audio and writing groups, and today there was a woman who was distressed because she had just moved, and in the move her laptop was totaled and she lost her backup flash drive with her novel on it. Apparently tech specialists told her they could not recover the laptop drive. 


Let's have a moment of silence... 


Keep three backups of your work, with one of the backups being in a separate location. Now, this is easier for writers since they are working with far smaller file sizes, but even when you're writing music, you seriously need to take this into consideration. It is not unknown for both primary and secondary backups to fail around the same time. Nor are fire and theft unusual. 


(TL;DR --> At home you want a minimum of your main, active drive(s) and a good backup of each. Use a proper backup program for this, such as Carbon Copy Cloner. For your tertiary drive, there are cloud services that can be used for backup, but you will have to pay, probably in the vicinity of $10/month depending on who you go with. Be careful of sync services vs real backup services. Finally, for long term storage of digital medium, optical discs still seem to be the best, meaning they are strongest and hold data for the longest amount of time. Just remember not to keep all three of your backups in the same location.)  


Cloud services are a choice. Again, with writers it's an easy choice, but if you're working with massive amounts of data, you might end up having to pay for extra space in a cloud service. One thing that I do is retire tertiary backups after a while, moving them into hard storage. If it's something you won't need to access very often, like a mix you are done with, Amazon's cold storage is a not-too-expensive option. The cheapest option is probably having an archival backup drive at a family member or mate's house. Heck, you two could swap drives, helping each other out. That tertiary backup drive becomes an archive that you pick up only when you are ready to retire more things to the archive. Of course, even that's not perfect. You'll want to copy/replace that drive every few years. (Don't forget the crazy pace at which tech changes. Personally, I think this is making it harder to keep things long term. Remember physical photo albums? I've got some from 70-80 years ago. How many digital photos have you and your friends lost over time?)


Anyway, I have my primary drive and backup drive at home. For smaller writing projects, and for larger audio projects (that I am currently working on), I keep them in a cloud service. But... I don't totally trust cloud services, either, so I have an automated script setup to copy one cloud service to two other cloud services. 


Setting up a bi-weekly automated backup script on a Mac

Easy peasy. 

  1. Open automator from your applications menu. (cmd-A for applications menu)
  2. Choose "Calendar Alarm". 
  3. Under the Library menu, choose "Files & Folders". 
  4. From the new menu that opens just to the right, click "Get Specified Finder Items" and drag it into the empty window area to the right of that. This is the pane where you construct your Automator task/script. 
  5. You'll see your "Get Specified Finder Items" task in the open area now. Click "Add" and find the folder or file you want copied. Add it. 
  6. Back in the Files & Folders menu, select "Copy Finder Items" and drag that to the open area in the right, under the "Get..." task you currently have there. 
  7. Click "To" and navigate to where you want the files copied. Click "Choose". 
  8. Tick the "Replace Existing Files" checkbox. 
  9. Click the Run arrow in the upper right to test it. 
  10. Save the automator script (cmd-S to save). It will now run the script from the Calendar. Calendar I believe also opens at this point. 
  11. Go to your new Calendar alarm scripted task in the Calendar and make sure it is set for the correct time and date that you want. Then click "Repeat", and set it for how often you want it. I run the script twice a week. (Select "Weekly", then in the Calendar copy the event and paste it into a new day.) 



So again for smaller files, I have my primary drive, my backup.... okay, okay I have two hard backups... one is via Time Machine, and another is via a piece of software I have called Carbon Copy Cloner, which is worth every penny. Then for the smaller files, I also keep them in a cloud service, with a script to copy them to two other cloud services. 


For larger files, I have the primary backup and two hard drive backups as mentioned, but I will sometimes keep active projects in the cloud services, too, as long as they don't get too big. I also used to keep an archive drive like mentioned above with my brother, but he lives halfway across the nation now, so we don't do that too much anymore.

Now that I'm write-thinking about it, I think I'll ask a friend or two if they want to swap archival drives. 


Yeah, so -- backup your files! At least three copies, and keep one in a separate location. Having your main drive and a single backup drive is not proper backup! Three is the magic number! 



Hard drives are getting smaller, and cloud services are getting cheaper. Here are some common larger options, with some of their pricing plans (not all).

(Please note that these services may not work if you are intending to cloud-backup an external drive. Check each one individually for cloud backup as opposed to cloud sync.)

iCloud - $1/month for 50GB, $3/month for 200GB, $10/month for 2TB

Google Drive - $2/month for 100GB

Dropbox - starts at $8.25/month for 1TB


Here's an article from Jan 5 of this year, reviewing some of the top cloud services for music.
Here's an article from Aug 2017, talking about applicable cloud backup services for backing up external drives.



This article says that studies done by France and the US Navy have confirmed optical discs as to be one of the better modern choices for archiving

It may be reasonable to buy a quality external optical disc drive if you don't still have one, and use it to burn copies of your completed projects and general software to archive. 






Blue Hawk Down

Studio Chai is currently down as we mourn the loss of our dear Svetlana. She was a good hack. We learned a lot with her, but she just didn't jive with an upgrade, and we're having to let her go. RIP, baby. RIP. 

I'm currently awaiting new parts, which will take some time, and expect to be back online in a couple weeks. Until then, er, uh, na zdraví I guess. :P

Btw, the Instrumentals Contest is up this month. This outlying contest on the Muse never really drew that many people, but was a recurrent love with a few back in the old Muse's Muse site. Not sure how it'll go on the new Muse, but it's up and available for any who want to take advantage of it. It's open to all entries, and is a good place to try out non-lyrical pieces you've been working on or have already finished that don't fit into the existing contests. 


When you're recording solo at home, it becomes more useful than ever to map out your song in your DAW, at least at some point. If the song hangs in the same time/tempo throughout, it's easy -- set the time & tempo (or not), and press record. Any markers after this are just for organization's sake. If, however, you change time signature or change tempo during the song, it'll probably be a good idea for you to map that out before recording, especially if you are adding VSTs later, or syncing effects, or exporting data to another DAW... if you're doing anything that requires any syncing, you'll want to do this.†


First things first, get pen and paper or open notepad, and write it all out. You probably have some of this done already -- maybe lyrics written down and organized, maybe chord charts. Now it's time to count measures, note the time signatures, and note the tempo with any tempo changes. Logic has an internal note-taking feature that I love. It allows you to both make project notes and notes on each track. This is incredibly useful. If there's any drawback, it's only that if for some reason you lose the ability to open that Logic session (if, for example, you move to a computer without Logic), you'll lose your notes. So it may be a good idea to copy your notes to notepad. Up to you. Ableton doesn't have this feature at all, and while I did find a way to fake mini notes inside ("extended names" you could say), it's highly inefficient, so I just keep a notepad file in the session. 


Tempo Changes in Ableton and LPX

One of the things I really appreciate about Ableton, and indeed one of its shining features, is its built-in and on-by-default warp ability. Essentially Ableton automatically tries to find the tempo and transients of any audio you give it. This is excellent when it comes to changing tempo. You see, in Logic when you change tempo, all your MIDI instruments speed up or slow down, but your audio files stay the same, at the old tempo. If you want to modify your audio files, you have to either screw around with the flex audio settings and do it all manually, or you have to re-record. Not so in Ableton. If you change the tempo in Ableton, all of both your MIDI and audio files will speed up and slow down according to the tempo changes. This is a fantastic feature that can really save time when you're going through drafts. 


That being said, changing tempo in the time line was a PITA. 面倒臭かった。In both LPX and Ableton, tempo will typically be worked via automation modifiers, but while LPX works in whole numbers, Ableton practically insists on decimals. It's incredibly annoying. You want that section in 125 bpm? Good luck. You'll probably end up with 124.32 or the like. This is not a big deal if you're staying in Ableton, but if you want to export files to lock to another piece of software, it might be somewhat problematic. And even if not, ffs, why can't I just have a whole number? Perhaps I'm nitpicking, but it does make a difference when going cross-DAW. 


In both DAWs, you can click on the tempo automation line to set mod points, but that's where the similarity ends. 


In LPX, once you have a couple of points designated, you can grab the bar between them and by default dragging it changes the tempo by whole numbers. If you want finer tuning, as in decimal tuning, just hold down control while you drag. Easy. (If you can't see the tempo automation area (which is off by default), you need to click the global track button, which is just above all of your tracks. Look and you'll see a plus for "new track", then another special plus symbol that also adds a track... and then a bit more to the right is what looks like an arrow in a window. That's your huckleberry.) 


In Ableton... after you click to create two points, if you click on the bar between them, that will just create another point. But if you move your pointer close to the bar, without touching it, you'll see the bar change color. (Purple on my comp.) That's the signal that it can be dragged as a bar. But you'll have that issue I spoke of where it changes the tempo by decimal degrees. You can hold down command and it'll kinda jump, but it'll still be decimal degrees. 


So... I have found two workarounds, neither of which are great. 


1) In arrangement view (this all has to be done in arrangement view), click the focus marker to where you want the tempo change to begin. Then in the tempo zone on the transport, type in the tempo that you want. This will move the temporary dotted automation line to your desired tempo. Now, to actually make that tempo change, you have to record. That's right, you have to hit record with automation armed. (By default, it is armed. It's the button in the transport bar to the right of the plus and to the left of the left arrow. It looks like a couple of boxes/points connected by a little line. More on this later.) So, you essentially have to record all the automation changes in. Make sure your audio/midi tracks are not accidentally also armed in this process. 


2) The second is by utilizing the draw tool. You can access the draw tool by right clicking in the automation window and turning it on in the sub menu. Now, this will pretty much operate in the same funky decimal fashion as before, though according to your automation quantize settings, which means it will change tempo a designed block/section/note at a time. But... on the right in your master track settings, if you select "song tempo" in the automation area, it will open up a couple of range boxes, which default to I think 20 and 200. If you change this to a very small range, or more importantly, change that upper range to the actual tempo you want, you can then use the draw tool to pencil in your desired tempo as long as you draw at the top of the automation area. Not exactly elegant, but it works. 


Okay... it's important to mention that Ableton has automation armed by default. I pointed this button out above. This means that if you move dials around while recording, Ableton will imprint those on the automation line. This can be annoying if you, for example, just want to raise the volume across a track. Click the button to turn it off. 


Now... this is something I really don't like... even with automation disarmed, if you go to an area where you want a different tempo and manually type in the tempo, this freakin' program will temporarily move your entire project to that tempo, complete with dotted line as if you're about to record a tempo change. This complete overrides all of your previous work, and you'll have to hit the "automation undo" left arrow button to go back to how you had it. 超面倒くせー。I see this as a leftover from Ableton's DJing days. 


Time Signature Changes

Thankfully, this is easy in both DAWs. 


In Ableton, you can right-click the upmost flag in the arrangement (actually looks like a half square) after you've moved it to where you want, and the sub menu will give you the option to "edit time signature". Simply type in the new time sig using a divisor / bar. 


In LPX, quite frankly after opening the global track settings as talked about above, you should see how to change time, change tempo, add markers, and more. Simply put the marker where you want the change to be, and in the time sig section, click the plus sign and type in what you want. Easy peasy. 


Adding Markers/Locaters

In Ableton, when you have the focus line where you want the marker/locator, in the menu above select Create > Locator. 

In LPX, when you have the focus line where you want the marker, click the plus sign in the marker area of the global tracks view. 


For both, you can rename markers/locators by right clicking and choosing "rename" from the menu. 



Blpffft.... what a boring (but important) post... The reason I wrote this post was because Ableton is so counter-intuitive when it comes to making tempo changes. I wanted to make a record of what worked, and then along the way thought I would compare it to LPX. 



† With many DAWs, you can record freely and then try to go back and get the DAW to recognize tempo. This is not so bad if you're recording drums, but if you're recording something that has muddier transients like an acoustic guitar, it can be pretty sloppy. Personally, I think it's better to just map tempo beforehand. It helps to know your song well before you record. Granted, you might just be drafting, which means you'll rerecord later, but that's part of the artistic process. 


V7 Display of Power

(Also known as "Functional Harmony in Extended Dominant Patterns")


You know, as a kid I didn't care for dominant chords so much. When I tooled on the piano, I stuck pretty much with major and minor patterns, and when I was growing on the guitar, I did a lot of rock and metal, some folk... but not music where that b7 really stood out. 9ths, yeah. Straight dominants, not so much. And when I learned basic dominant shapes on the guitar, I didn't like them. 


Silly me. Dominants rock. 

Heck -- blues, swing... there are a number of butt kickin' tunes that hang in dominant movements pretty much the whole time, if not THE whole time. Part of the magic of ye ol' 12 bar is that distinct sound of blues minor soloing over those dominant shapes.


Alright, dominant chords resolve excellently to the tonic of the key. V7 to I, G7 to C, a match made in harmony. 

In doing that V7 to I, you can easily preclude the V7 with a ii. For example, D-7 G7 C, ii-V7-I. 

Now, that ii can be precluded with its own dominant chord, acting as if the ii is the root. The dominant of D is A7. 

Now we have A7 D-7 G7 C.  <-- pay attention to that if you read these chords backwards, you're moving through 5ths. 

The A7 in turn can be preceded by another "ii" chord (operating with the A as the dominant in the key of D). 

This gives us E-7 A7 D-7 G7 C. And you know what, screw it, let's make all these new chords dominant. Because we can. Because it works. 

E7 A7 D7 G7 C  <-- This is an extended dominant pattern, known as the II-V7. You can keep this up as long as you like, stretching backwards by 5ths to make each new chord. (Of course keeping the ii chords as minor(7ths) sounds good, too!)


You can also move downwards in half steps (minor 2nds) with your dominant chords, and this also is called a II-V7 extended dominant progression. I'm... not exactly sure why yet.


Now you can do all this ad infinitum, going hamster wheel to your heart's content, but more likely you'll use these in short phrases -- maybe of only 2 or 3 chords -- in the midst of whatever else you might be doing. As always, feel it out and experiment. See what you like. 


One last thing -- we can't have a conversation about dominants and substitution without bringing up tritone substitution

I learned about tritones from Slayer, of all things. The ol' Devil's Interval, the tritone is 3 exact steps, or 6 half steps, placing it in the middle of the octave. I generally look at it as a b5, but of course #4 is fine if you're a potato. I mean potato. You know, 'cause potato pota... oh never mind...


So tritone substitution is when you take a dominant chord -- let's go with G7 since that's how we've been rolling -- and substitute another dominant chord built from the tritone of the root of the first chord. The tritone from the root of G7 is Db, so this means that Db7 is a potential substitute for a G7 chord. And if you look at the notes of the chords: 

G7 = G B D F

Db7 = Db F Ab Cb 

You can see that they each share two notes: the B and the F. And you know what? B to F is a tritone. 

Mind... blown..... whoooo



In Basic Chord Substitution, we mentioned parallel substitution, also known as modal interchange. An example of modal interchange is using the bVIIMaj7 chord in a major/ionian key, a chord which could be considered borrowed from the dorian (b3, b7) or mixolydian (b7) scales, for example. 


bVIIMaj7 in use (BbMaj7-G-C) 


Take a look at this chart of the modes of C harmonized in 7ths: 



Basically, chords sharing a column are ripe for substitution via the modal interchange method. There are of course many more scales than this, and after a while quite frankly it starts to mean that just about anything can become just about anything. That's cool. But it's nice to have reference points, islands plotted along charts of the journey, if you will. 


Here are some popular modal interchanges not covered in the chart: 

IMaj7 to I-6 or I-Maj7 (IminMaj7)

IVMaj7 to IV-6

V7 to V7(b9)


Again, there's really no end to this kind of discussion, and it comes down to experimentation, to what feels/sounds right to you.


Basic Chord Substitution

You could spend a lifetime delving into chord substitution and reharmonization, but we're going to talk about fundamentals here, the basics. 



First of all -- major, minor, and dominant chords. 

Feel free to substitute any one of these for another to see how it sounds. Experiment. A sudden dominant chord where not expected, or a sudden shift from major to minor or minor to major, whether mid-phrase or when moving into a new phrase, sometimes can bring out very nice harmonic qualities. 


Take this basic progression: 

G  Bm  Em  C   G   (I-iii-vi-IV-I)

Adding a single dominant chord changes the flavor, making the progression more playful: 

G  B7  Em  C(maj7)  G    (I-III7-vi-IV-I)   


Listen: Ukulele Sub


Or in this progression: 

Em  F  Am  Dm  F  Am  Em  Am   (v-VI-i-iv-VI-i-v-i)

Making a couple dominant and minor substitutions gives it a rather different feel: 

Em  F(maj7)  Am  D  F/Dm  Am  E7  Am   (v-VI-i-IV-iv-i-V7-i)

(It might be easier to consider the above longer progressions as groups of two, i.e.: v-VI,  i-IV,  iv-i,  V7-i)


Listen: Piano Sub


Or listen to "Norwegian Wood" for a great example of a shift of harmony from major to minor. These sometimes can bring about a total change in key (parallel modulation), but also can simply be temporary substitutions within a phrase. It's up to you.


It's worth noting these types of substitutions are sometimes called parallel substitutions (or modal interchange) because the different chords can be found in keys parallel to the tonic. 



Next, let's consider diatonic families, using everybody's favorite, the C Major/Ionian scale. The scale itself is good ol' C D E F G A B C, which harmonizes to:

C  Dm  Em  F  G7  Am  Bº/Bm7b5†  C

In other words, harmonizing a major scale gives us:  

I  ii  iii  IV  V7  vi  viiº  I

(It's also pretty common to harmonize in 7ths, giving Cmaj7  D-7  E-7  Fmaj7  G7  A-7  B-7b5  Cmaj7) 


There are three families in the diatonic scale: 

The tonic family -- I, iii, vi

The dominant family -- V7, viiº (VII-b5)†

The subdominant family -- ii, IV

(It is worth noting that V7sus4 is another common substitution chord considered to be a part of both the dominant and subdominant families.)


A chord in a family is generally an easy substitute for another chord in the same family. It's pretty easy to see the reason behind this. Looking at the notes in C major, C  E  G, we see that the other chords in the same family each share at least two notes with C. Em = E G B, and Am = A C E. 

(By this logic, although IV and vi are not in the same family, they sometimes can substitute for each other.) 



We should also point out that when you harmonize a minor scale, things can be a little different. First of all, the fifth chord will naturally be a minor, e.g. an Em in the A natural minor scale. It is completely normal to turn this into a dominant chord, E7 in our example of A. It fits the harmonic/melodic minor scales, having the leading 7th tone (G#), and gives a nice, strong dominant to tonic sound. This also means that you can harmonize the minor scale as such (going into 7ths for this one): 

I-7  II-7b5  IIImaj7  IV-7  V7  VImaj7  VII7†


Even in this case, the above "rules" apply. Like mentioned before, a general way to consider it is by looking at what chords share notes. For a good substitution, you want at least two notes to be shared. There's a lot more that could be said on this, but let's keep it simple. 



So, this is Substitution 101, and already there should be a wealth of potential here. Being aware of voicing, experiment and see what works for you. 


†There are several chord "nomen"clatures. A major 7th chord can be written as Cmaj7, CM7, C∆7 (if tonic in a harmonization, it can can be written as Imaj7); a minor 7th can be written as Cmin7, Cm7, C-7, I-7... when written as just triads, capitals generally are major, and small letters are generally minor -- III vs. iii


Feedback loops between effects can make some really wonderful sounds sometimes, and it's quite easy to do.

Here's an example of it being used. It's being created with delay and reverb and comes in as a ghost-like sound that follows the cello. It's first strong around the 1m mark: 




**Warning: Put a LIMITER on your effect tracks and your master track when you experiment with these, at least until you learn how to ride them, so as to not blow anything.**


Feedback loops are simple. I pretty much always do these between a delay and a reverb, or two delays. You essentially feed your track single into one of the effects as always, then you feed that effect into the other effect, and then you feed that second effect back into the first. Now, when you do this, you essentially have to treat it like an instrument, keeping a hand(s) at all times on those send controls. If you don't, the feedback will get out of control and all you'll have is noise (that might blow your ears/speakers). So the key is to ride the sends, pulling sounds out of the feedback. 


There is one thing that Ableton does in this regard that I don't remember other DAWs doing, and that's that it lets you feedback an effect into itself. It's setup to do this by default because of the way it auto-inserts send knobs on all tracks. Interesting. 


In the example clip below, I have a loop of a kahon and two return tracks setup to receive sends. This is actually Ableton's default template. Send A goes to reverb. Send B goes to a delay. Notice the first thing I had to do was enable the sends on the aux/return channels themselves. Once I did that, I could play around with looping the effects within themselves and with each other. This is personally a favorite effect of mine as you can get some great ghostly textures out of it. 


(FWIW, you can also do this with hardware, but again be careful not to damage anything.)




The autopan plugin in Ableton can be used to create a gate or staccato-like effect. To test it out, I loaded up some of Ableton's stock loops, and then put the autopan on one track. Essentially when you set the phase to zero, it gets its gate-like qualities. You can set the LFO type to song tempo, and then adjust how fast or slow you want it while it auto-syncs. Once you have it in this mode, the Amount knob acts like a Wet/Dry knob, and the Offset acts like a to-tempo fine tuner. Interesting, all in all. 



Randomizing Clips

So, I've mentioned that I picked up Ableton Live on discount, and have been tooling around with it. In the past, I was turned off because I always saw people using the Session view in Ableton, which seemed to have been designed with DJs and loop artists in mind. I'm a linear writer, so that doesn't really work for me. However, Ableton also has the Arrangement view, which is the linear timeline that we know of in your normal DAW. Not only can you move back and forth between these work views just by tabbing, but if you click and hold something you've created or loaded into one view, and then tab into the other, it will take that clip and put it into the other view. It's really easy, and actually pretty cool. 


Anyway... Ableton got my attention again with its inclusion of Max for Live. If you don't know about Max for Live, this basically integrates Cycling74's Max plugins into Ableton. Max is a gui programming interface that is popular in audio and video circles. Having Max for Live gives you access to a wide range of plugins. It also means you can design your own plugins. Again, pretty cool stuff. 


So, Ableton is designed to allow creators to really experiment. I'm finding a lot of things that it can do that either other DAWs can't, or that they can, but not as easily. Now this clip randomization thing I'm talking about today is not necessary, and indeed like the Device Randomizer, you probably won't use it often, but it is one of those ways you can tool around with ideas and maybe get a happy accident or two. 


Basically, you load up a number of audio clips on one track in session view, highlight them all, and using the Launch window in the clips area, tell Ableton to randomly jump through the clips in a manner that you assign. This is what the Launch box looks like, here in the middle:


If you don't see the Launch box, then click the "L" in the lower-left hand corner. (See above)

In order to make Ableton jump randomly through pieces of your audio clips, you'll need to highlight all the clips in the track (and note, if you have a space between the clips in session view, Ableton will ignore the clips below the space), then in the Launch box, you need to set the second box down under "Follow Action" as an asterisk. The asterisk is the wild card, telling it to go random. The first three boxes under Follow Action tell Ableton how often to switch clips, in bars, beats, and sixteenths. That's very important. I experimented mostly with eights and quarter note changes. The last two boxes underneath (here set to 1:0) tell Ableton how often (in a ratio) to do the actions above, but are not important for what we're doing.


The legato button is however very important. If you do not have this engaged, Ableton with start each clip from the beginning every time it jumps. However, if legato is engaged, Ableton will launch the next clip during what part fits where you are in the measure. 


Those are the most important bits. I have it set on repeat to make sure it keeps playing, and Quantization at 1/2 measures, though I don't think that is so important. 


Here's the test run that came about from this. I had a bunch of similar drum loops, and coupled this with a track full of random vocal samples. I also played around with envelopes and automation a little. This was all just to learn how it worked: 




If you want to learn more about the Launch window, here are the pages on it in the Ableton manual.


If you're interested in Ableton Live, you can download it and try it for free for 30 days.






I saw this book on “Jazz, Rags & Blues” for cheap and picked it up. It’s an easy book, rated as around intermediate level, but the songs are well constructed and great to improvise on. Although piano was my first instrument, I moved to guitar in my pre-teens and never developed piano as well as I would have liked. That being said, recently I had wanted to get more into swing and jazz piano, so this book was an easy choice. (I also used to love playing rag, with a compilation of Scott Joplin’s songs as one of my old favorite piano books.) 


Time to time I like to analyze the chord structure of songs that I like, to see what’s underneath. Today I want to talk about one of the songs in this book, called Red Rose Rendezvous. 


There’s a video of the piece being played here. It's being played it at what I would call a teacher’s pace. Personally, I like to vary the dynamics and tempo; but again, that’s one thing I like about the book — the songs have a great, easy structure that beg to be built on, improvised. 


The overall structure of the piece is ABABCABB. 

The song moves in 3/4, but the A section has a distinct pull towards being 5/4 a la Take Five. 


The A section is: 

Dm  G  Dm  G  (repeat)  A+   A

So basically a i-IV-V-i 

I do like that A augmented at the end. It’s a good reminder of simple, effective embellishment. 


The B section switches gears in that, while the underlying rhythm pattern is the same, it’s pedaled, giving a distinctively different feel. Let’s look at the individual triads in that descending section: 

D F A — Dm

C F A — C/F

B F A — B/F6  or  Bø7

Bb F Ab — Bb/Fm6  or  Bb7

A D F — A/Dm 

A — A (implied major)


So… we could take the route of saying this is just a Dm followed by various embellished F chords, going back to the tonic chord and then the fifth, A. This would sum up as a i-III-i-V, basically. 


However, we could consider a couple of those F chords as Bs. F and B(b) have a subdominant/dominant relationship with each other, which explains why they can sound so good next to each other. And if we look at the last three chords, on their own they make a bII-iv-I progression, the tonic of which (A) is the dominant to the key of the song as a whole (Dm), making it easy to slip back into Dm for the next A section. 


Cool beans. 


Section C opens up with what appears to be a modulation to the parallel major, F:

Gm  C7  F  Dm  D/Gm  C#/A7 

ii - V7 - I - vi - ii - III7 

The A7 at the end is a III7 to F, but acts as a V7 to temporarily move back into the key of Dm, before moving back into F. Then the phrase repeats with some changes at the end: 

Gm  C7  F  Dm  Gm  Gb+  F/B  E/C#º(7)

I’m not going to get into those final chords. Suffice it to say, after the second Gm, there is a descending bass line with a bit of harmonic scale tonality en retard and a hard modulation back into Dm as we enter the A section for the last time. 


The song ends on a variant of the B section. I kind of like to play with the final chords here. Instead of ending with Dm  C/F  Bø  Bb7  Dm7, I kind of like to go Dm  C/F  Bmaj7  Bº7  Dm7. It does modify the “perfect” descending bass line, but I like the tonality. 


And finally the song ends on its original tonic, Dm, which is dusted with a high and light Dm6/9. Very nice touch. 


Anyway, very cool song, very easy to analyze and improvise on, and a very cool book altogether. I think I’ll talk about other songs in this book at a later date. I’ve really come to appreciate her work. 


Jazz, Rags & Blues by Martha Mier





Device Randomizer, Part 2

(EDIT: Think of the Device Randomizer as a response to the MIDI randomizer. It does the same thing in concept, but randomizes plugin parameters. That being said, what I haven't done, but absolutely should do especially now that we've established how to configure/setup 3rd party plugin parameters in the device view, is setup the Device Randomizer to randomize settings on VST instruments. This could be a cool way to find new sounds. Up till now I've just been messing with effects and pass-through modelers.) 


While I don't think this device really needs two blog posts, I'm putting another one up because I decided if I'm going to learn how to use it, I should really learn how to use it. Thankfully, it's a pretty simple device. 


(**Note: always put a limiter on your master buss and/or at the end of your track chain before doing these kinds of crazy sound design experiments.**)


Like I mentioned in part 1, the Device Randomizer does exactly what it says -- randomizing the settings on whatever device you want. 

Now, on part 1, I had it mostly just randomizing resonators and grain plugins chained to field samples, but this time, I've got a drum loop (Afro-Caribbean) running through Corpus, a physical modeler, and a Waves flanger. See the video below. 


The Device Randomizer has three sections: 

  1. Auto is where it randomizes things continuously. When you turn on Auto, the Device Randomizer will keep changing all the parameters you tell it to. How it changes those parameters depends on the settings. Click the little black triangle in the upper right to expand the Device Randomizer (already expanded in the video), and access those settings. Smoothing and Curve tell it how fast to move through changes. If you play around with these, it'll go staccato on you. Depth tells it how far to go given the other settings. Notice there are frequency settings here, too. These act as time LFOs to the parameters on their right. If you click the blue Freq buttons, it changes to "Sync", and instead of LFO-type sweeping actions, you can set time parameters locked to tempo on a mega range of whole notes to 128th notes. (Note that Auto will not work with some plugins, essentially just becoming another Trigger. Such is the case with Corpus--the parameters change, but nothing happens until I stop it.)
  2. Trigger is your one-shot randomizer. Kinda like a roll of the dice. Its two settings are the percent degrees of Offset and Randomization. 
  3. Edit is important because this is where you tell the Device Randomizer what parameters to affect. A lot of the time, you will not want it to go nutso on every parameter for your effect/device, so you specify what here. Note that mapping the Device Randomizer to filters can yield interesting results, and this is a case where you would probably want to lock down what is affected. (The blue box in the corner adds "Gain" as a parameter to be randomized, if you really want that.) 


What if you want to use the Device Randomizer on a 3rd party effect that doesn't come with Ableton? 

Well, in Ableton when you add a 3rd party virtual instrument or effect, clicking the little black triangle on the effect in device view will expand the device and allow you to add parameters to the device in the rack via the "Configure" button. Some 3rd party plugins are already setup for you. The Device Randomizer will recognize anything configured here. Notice the MetaFlanger I have is from Waves, and that in the device view for it, 16 parameters have already been defined. 


So, here's the weirdness: 



Device Randomizer, Part 1

The Device Randomizer is part of the Max for Live kit. It's weird, to sum it up. 


Essentially, you put it in your audio chain, click the "Map" button, and then click the name bar of whatever effect you want it to go nuts on. Once you do that it will, you guessed it, start randomizing the parameters of that effect. (Thankfully the default is to do it smoothly.)


Notice I have two of them in the picture.

  • The first one is set to "Trigger". That means that I press the randomizer button once, and it randomizes the setting once. This is a cool way to come up interesting sounds on a bored day that you might otherwise not have come up with on your own. And it's fast. 
  • The second one just has a big blue block that says Randomize, and it's set to "Auto". When you select this, you're basically telling the randomizer to go crazy. Fourth of July, baby. It will continually change (pretty much) all the settings on the effects device it is bound to. Smoothly, thankfully; though the blue drop down arrow on the upper right of it gives access to settings that will modify how it moves through changes. 


So what the heck did I do with this thing? Made a bunch of crap, mainly, hahaha. There was one texture I kept, but really I was just messing around with Ableton, trying to see what all these tools do. (And there is a TON of them. I will never learn them all.) 


Here's the keeper texture. It was made by selectively randomizing on trigger, then resampling and applying a few other touches: 

Dark Texture


Here are some chaos textures I probably will never use, but are interesting in their own right. (Honestly I kept them just to post here so you guys can better hear active randomizers in motion -- note that I have randomizers on a resonator each, and then either a "swirl" effect or a "raindrop" effect, but you can use them on anything. (The first sample has also had the granulated pitsch shifted out of it.)) For both of these, I just let the randomizer go crazy. The first one is using the same sample as above, the second is totally different, and also has a totally different effects chain: 

Random Mush

Saturn Weirdness

EDIT: Ooh... open the two mushy chaos crap links above in separate tabs and play them together. Much more interesting. 


Yeah, weirdness, but the point is to know that the tool exists and how to use it. It's a decent plugin for finding surprises or just screwing around with at the end of the day when your brain is turning to mush. Like now... 



Screen Shot 2017-12-16 at 10.40.59 PM.png

I got this piece of software practically for free after a back and forth with iZotope. Love those guys. They gave it to me bare bones, though. Anyway, it's a really strange and interesting piece of software, but I was having a hard time getting polyrhythms out of it. Here are my notes for any of you who may have it. 
(always remember you can flip the dominant pulses)
1- 1x speed, 12 steps, beats on 1-5-9
2- 2/3x speed, 8 steps, beat on 1-5
1- 1x speed, 12 steps, beats on 1-5-9
2- 1x speed, 12 steps, beats on 1-7
1- 1x speed, 16 steps, beats on 1-5-9-13
2- 3/2x speed, 24 steps, beats on 1-9-17
1- 1x speed, 12 steps, beats on 1-4-7-10
2- 1x speed, 12 steps, beats on 1-5-9
1- 1x speed, 20 steps, beats on 1-5-9-13-17
2- 1x speed, 20 steps, beats on 1-6-11-16
(Cut #1 down to get variations with 3 and 2 beats. #2 will find its pace.)
1- 1x, 28 steps, beats every 4 steps
2- 1x, 28 steps, beats every 7 steps That might be all with this one…
Curious to learn more ways to make polyrhythms by expanding/shortening the bars of certain instruments and changing their relative speed...
Fun Fact:
In harmony, 3:2 = perfect 5th, 4:3 = perfect 4th, 5:4 = major 3rd
(7:4 is a “harmonic 7th”…. While we’re at it, 16:15 is a m2, 9:8 = M2, 6:5 = m3, 8:5 = m6, 5:3 = M6, 16:9 = m7, 15:8 = M7)
Also…. 2 ways to get 6:4
1- 3/2x, 24 steps, beat every 4 steps
2- 3/2x, 24 steps, beat every 6 steps
2- 1/2x, 16 steps, beat every 2 steps (more like a hybrid 6:4 & 12:8)

Endless Modulating in 3rds

I was working on a piece that had a section built on Gm-F# (which I'm seeing as either a bii-I or vi-bVI(-I)†), but needed an extended finish which kept that kind of pace. I realized it was pretty ripe for a looping modulation in 3rds. 


Essentially, you modulate to the 3rd of the chord, going down, then play whatever you have in mind (obviously what you have in mind is important), then modulate down again on the consequent 3rd. This is really easy to do with this bii-I like movement, and basically turns it into a modulation cycle that goes back to your original key after 4 modulations. (Cause, you know, three times four...) 


So what happened was: Gm to F#, mod to Bbm to A, mod to Dbm to C, mod to Em to D, mod back to Gbm to F#... 

Gm F#

Bbm A

Dbm C

Em D#

Gm F#

Of course, you need to mind your intervals while descending. In my case, I went for the simplest changes. 


Here's a (sloppy) clip showing you what I'm talking about. (Yeah, yeah, it was a political piece that I never finished. I don't normally do that. Write political pieces, that is. It wasn't my intention when I began. There's a story to how it happened.)


Note also that if you just take a chord and keep modulating down an actual third interval (from root), you get something reminiscent of neverending doo-wop progression. 


† vi-bVI-I is interesting because, say you're going Gm-Gb-Bb (2nd inversion), it's a neat little way to have a simple semi-tone descending bass line to the tonic chord. Here's a little clip of this in C (Am-Abmaj7-C6).


In Ableton

— Right click on the clip, and select one of the following:

**Convert Harmony to New MIDI Track

**Convert Melody to New MIDI Track

**Convert Drums to New MIDI Track



Converting a melodic audio file to MIDI in LPX

— Double click the audio clip to open edit mode

— Turn on Flex

— Change Flex to Flex Pitch

— From the Edit menu, choose “Create MIDI track from flex pitch data”

Here's a video of it being done (with bugs in the process (what version of X?))


Converting drum audio to MIDI in LPX:

Essentially you use the “Replace or Double Drum Track” in the Track menu, which creates an EXS24 instance, and from the dialogue box you set threshold values defining each instrument you want Logic to look for. Here's an official tutorial page to do this



Par contre... 


To convert MIDI to audio in LPX:

-- Right click on the MIDI clip and select "Bounce in Place" (or alternatively "Export as Audio File")


To convert MIDI to audio in Ableton

-- Right click on the clip and choose the option to Freeze

-- In arrangement view (drag it there if it's missing), create a new audio track with cmd-T (if you don't already have one for this purpose), then drag the frozen MIDI clip to the audio track; Ableton will put it there as audio



The purpose of this blog entry is not only to show how to do these things, but also to show that where one function is easy in Ableton and harder in LPX, the reverse function is easy in LPX and somewhat harder in Ableton. 


EDIT: Other DAWs such as Cubase and Sonar can also do (some of) this. So can some plugins like, I think, Melodyne. I'm just focusing on what I'm using now. (I would look up how to do it in PT, but right now Svetlana isn't jiving with PT...)