The PadKontrol has one button labeled ‘Message’, which you probably never use. Holding down the Message button and pressing one of pads 1-8 will send a MIDI message to the software or whatever the PadKontrol is hooked up to. The first three pads send Panic, Note Off and GM On messages, while the next five can be configured to send any MIDI message. 

You’ll need the PadKontrol Editor Librarian software, which you can find on Korg’s website. It won’t work if you have your sequencer open while running the editor. From the MIDI menu, select ‘Edit Global Data’, and the Global Edit window will pop up. By typing in a hexadecimal code into the User Message field, you can get the PadKontrol to send whatever MIDI message, so we’ll have to find the hex codes for MMC messages. A quick trip to wikipedia brings up a list of MMC commands and their hex codes.  

An MMC message (that is sent to, or generated by, an MMC device) is: 

F0 7F deviceID 06 command F7  

The third byte is the Device ID.
The fifth byte is the command:
01 Stop
02 Play
03 Deferred Play
04 Fast Forward
05 Rewind
06 Record Strobe (Punch In)
07 Record Exit (Punch out)
08 Record Ready (Record Pause)
09 Pause
0A Eject
0B Chase
0F MMC Reset
40 Write
44 Locate/Go to
47 Shuttle

  

I don’t know what the deviceID is supposed to be, so I just used 00, and that seemed to work. The only commands I was really interested in were play, stop and record, but there are three different record commands. Through trial and error, I found that the Record Strobe (Punch In) command did what I wanted.  So, the record message looks like F0 7F 00 06 06 F7.  

 Then I typed the MMC message  into the box for the pad that I wanted to send that message and pressed enter to confirm. You can’t just type it in and click away, you have to press enter for it to remember. When I was all done, I pressed Transmit and tested it out.  Here’s a screenshot of the editor window with all the messages typed in.  

The MMC Messages

ZERO FREAKING CROSSING

What is ‘zero crossing?’ A zero crossing point is a sample in an audio file that is completely silent (-infinity decibels), or near enough. Always make sure your samples begin and end on a zero crossing point. If you don’t, there will be an excruciatingly annoying clicking sound whenever the sample plays. Edison has a handy declicking button that fades the selection in or out to silence.  

ABRUPT ENDINGS SUCK

A good drum sample will have a nice, natural sounding reverb tail on the end. Even if it’s a pretty dry sound, a bit of a roomy fade at the end will sound nice. If you’re chopping one-shots out of a loop of some sort, reconstructing the tail of the drum will always make it sound better.  

CHOP OFF UNNECESSARY SILENCE

Always delete any silence at the beginning of a sample so that it starts right before the first transient. Having a nice ambience at the end makes things sound good, but once the signal gets down to -70 or so decibels, chop the rest off. It’s just wasting space. It can be hard to tell where there’s silence when looking at the waveform. A spectrum view of the sample will show dead silence much easier.  

CHOP OFF UNECESSARY SILENCE

44.1 khz 16 bit WAVs PLEASE

I’ll take your MP3s, I guess, but they look ugly, and I have more than enough hard drive space.  

MP3s ARE UGLY

So I’m going to take a few snare drum samples from my excessively large collection and layer them together to make some new snare sounds. These samples will be destined for the FPC, so they’ll be of the obsessive veloctiy-mapped multisample variety. After they’re layered up, I’ll just pop them into Edison and pop them back out as individual wavs. Fun.

Layering Drums in the Step Sequencer

To begin, I just throw a bunch of snare samples into the step sequencer. I try to make the sounds ascend in loudness and hardness, so that soft snares are at the left and big loud ones are further on. Obviously, there’s a lot of volume automation and the pitch of some samples are varied between each step, so that it sounds natural. Each step should sound unique in some way, as the tiny variances between velocity layers is what adds realism. Alt-R will randomize both the pattern (which we don’t want) or the levels of one or more of the channels. I used this to lazily add some volume and pitch variations on some of the layers. When I’m setting up all these layers, I have the tempo set to about 60 bpm so that I can hear all the drum sounds in succession. Later on, when I start recording, I’ll lower the tempo so that each hit can fade out before the next one begins.

Reverb, Eq, Compression!

When I’m pretty happy with how the snares are sounding, I’ll put a bit of compression or reverb or whatever else might sound good on the whole deal. I’m using the compressor inside the Fruity Limiter. ‘Masterizing’ samples is kind of dumb.

Low end rumblies

When the compression and reverb or whatever else is sounding pretty good, I’ll add Edison after the last plugin and record the whole pattern. I set the tempo to 15 or so bpm, and there’s a bit of unneeded space between each hit, but we can trim it out later. While in Edison, I’ll just EQ out all the low-end rumblies that don’t do any good. Ctrl-N will normalize the whole recording, and we might as well get that out of the way.

Auto Slicing

Right-clicking the auto-slicing button will bring up a list of slicing options to add a marker to the beginning of each snare. Edison’s auto-slicing is usually pretty good, but it’s definitely not infallible. After slicing it up with dull auto-slicing, I went through and looked at each slice. Most of them were pretty much on point, but some of them were way off. Since I knew that every sound started on an eighth note, I set the tempo of the sample in Edison to 30, in the sample properties window, which can be brought up by pressing F2. Then I sliced it again with medium grid-slicing, and it all worked out great.

Renaming Regions

When we save all of these regions as individual wav files, each one will be named after whatever the name of it’s marker is. AwesomeSnare_Marker#6.wav is a bad name for a sample, so I’m going to rename them all to just numbers. Edison’s rename all function in the regions menu will do this. It’ll play the region and you type in what it’s name should be.

Silence

There’s a lot of silent gaps all over the place between these samples, and it would be a waste of space to keep them in. I converted the whole sample to 16 bits and used the spectrum view to delete all the silent parts. I could have kept them all in 32 bits, but that’s also a waste of space.

Export Regions

Now for the big moment, Export regions. I’ll just make a new folder where I want them to appear, and all my drum samples will be saved. Nice.

The EQ in Edison has some options in the dropdown menu accessible from the little arrow next the strange ‘Envelope’ label. There’s a few options for filtering out unharmonic stuff, based on the middle note. The middle note of the sample is set by right-clicking the darker grey ‘Title’ area in Edison, and selecting the middle note down at the bottom of the resulting window. This note will be the root note in a sampler, if the sample is saved and loaded into one. The root note is the note that must be played on the sampler in order for the note to play at normal speed, and when recording instrument multisamples, the note of the sound that was recorded. Edison’s EQ will use the root note to filter out anything that isn’t a harmonic of that note.

The EQ button and root note

Switching to the ‘Spectrum’ view by pressing S, you can probably see pretty clearly where your nice saxophone (or whatever) is. Edison has a lot of options for the spectrum view, one of the more important is that pressing N will switch between normal and ‘natural’ view. All this does is change the scale of the display. In normal mode, the frequencies are scaled linearly across the height of the window, but in ‘natural’ mode they’re scaled with a big fat emphasis on the lower frequencies. This is nice, as the lower frequencies are a lot more important. So, if you’ve identified whatever instrument you want to cut out, remember what it ‘looks like’ and click the EQ button.

A little snippet of a horn of some kind, or something. The arrow is pointing to some obnoxious 16 Khz whine, possibly due to a CRT monitor.

 The display inside Edison’s equalizer is different than the main view in Edison. Frequency is on the horizontal while time and level are on the vertical. There is a knob to adjust the scale. Hovering the cursor over top of the fundamental frequency of the sound that you want will display it in FL Studio’s little tooltip area. From here, it’s easy to set the root note of the sample to whatever FL Studio says in the tooltip, bring up the EQ window again and select an unharmonic filtering option from the dropdown arrow.

There seems to be two different pitches going on here, so I might do the whole procedure twice, selecting a different part.

The EQ window displays a neat pattern with all the little points, you click accept, and it might sound good. Mine didn’t sound good whatsoever, so I’m going to try it the labour intensive, manual way. All I’m going to do is copy and paste the whole sample, invert the phase on one of them, EQ out everything I want from it and combine them back together. Everything cancels out except the stuff I want to keep. I suppose I could just silence everything I don’t want in the EQ, but that would take longer, probably. The ‘Hold’ mode of the points in the equalizer are great for this, right click a point and select ‘Hold’. Hopefully, all the new points you make will also be ‘Hold’ points.

Hold

Well, this didn’t work very well for me this time, but that’s okay. Sometimes it works good.

In ye olden days, ring mod was done with a bunch of transformers or something. Plug pretty much anything into the two inputs, and martian sounds come out.  Complex chains of delays and reverbs and whatever else make a whole bunch of unpredictable, musically useless sounds. Woo!

These days, ring modulation is easy. Just multiply the two signals together. Strangely, it seems like most VST developers don’t really see the fun in doing ridiculously weird things with ring mod, so most of them just slap an oscillator on to modulate with the signal. There might be some different waveforms, variable pitch, maybe an LFO or two, but it’s always pretty much the same sound, and you’re left thinking about the wonderful things that could have been.

In frustration, I decided to make my own amazing ring mod VST,  in the very early morning. FL Studio didn’t support multi-input plugins at that time, so I made it with the same concept as FL’s crappy vocoder. Two different, hard panned sounds are routed into the plugin, and one modulates the other. Since ring mod just multiplies the signals, neither one is the ‘modulator’ or ‘carrier’, they both combine to form the output. With some clever routing, there’s funsound all over. I only wish that FL Studio would let me do crazy feedback stuff, but I guess that’s what virtual audio cables are for.

Easy.

It’s much easier to find a good balance between dynamics and loudness with parallel compression, rather than having just one compressor on the drums and spending hours trying to find the exact right settings. As a bonus, there’s a whole bunch of neato things you can do with parallel compression, especially with automation.

Like a lot of things, the one setting you finally decide on for the mix between squashed and unsquashed isn’t likely to be completely ideal for the entire song, unless your music sucks. Automating the level of the squashed, distorted drums so that during big, exciting parts of the track the drums become huge and massive and then laying off it a bit during calmer sections adds a lot of Good Things™. No one will probably notice, but it’s a lot of fun and it’ll make you feel good.

Depending on how it’s done, not all of the drums have to be sent to the squashed bus the same amount. I like to strategically send some drum sounds to the squashed bus more than the light n’ dainty bus so that the overall character of the drums can be changed, without switching the samples around or other time consuming things. Generally, I’ll send anything I think will add a lot of intensity to the drums more to the squashed bus. Usually cymbals, maybe hi hats and if I’ve layered two snare or kick drums together, I’ll squash one of the layers more than the other.

In FL Studio, I like to have a mixer track for each drum sound, and then send them all to two different tracks, compress one to death and sometimes widen the stereo or whatever else, really, and just make an automation clip for it.

Parallel Compression In FL

In Reason, I would have a Combinator with ReDrum or whatever I’m using for drums and a mixer for each drum sound. Putting a compressor on one of the sends and then automating the return level of that send would work pretty nicely. Each of the channels in the mixer can be sent to be squashed a different amount, woo!

Parallel Compression In Reason

That’s all. Go forth unto the world and make phatdrums.

Using this to re-arrange stuff could make some things easier, but it could also screw everything up and be frustrating.  

 The mixer tracks that individual instrument channels are sent to will change if you move stuff around, buuut multi-output plugins (or at least FPC) will not change which tracks the multiple outputs are sent to.  FPC’s multi-outs work by offsetting each pad’s output from the mixer track that FPC is sent to, in the channel settings window. That’s confusing. If I open up FPC’s channel settings window and send it to mixer track #2, and then set the output offset of a pad to 1, the output of that pad will be sent to mixer track #3. If I put a whole bunch of effects and stuff on mixer track #3, and then moved the fourth track one space to the left, the third mixer track would become the fourth, but the pad in FPC would still be sent to the third, and then I’d get mad because my drums sound bad all of a sudden. If I moved the fourth mixer track over two spaces, it would change the output of the FPC in the channel settings window to 3, and my drums would sound the same.

This way, it’s super easy to add another track next to the one where all the drums are sent and then do some parallel compression, or something. This was badly written, and I wish there were better names for things other than channels or busses or tracks.

Making your own cheesy drum machine sounds is easy, and you can use pretty much any synthesizer if you have a grasp of how it works. Pitch envelopes are used a lot to make drum sounds, so a synth with nice envelopes will be good. I’m going to use Sytrus, because that’s what I use for everything and its envelopes are amazing.  Let’s break from the usual large blocks of rambling text and break this up into sections on each drum sound. Looking at the waveforms of drum sounds and their frequency spectrums is useful to figure out what’s going on.

Bass Drums

A bass drum sound is just a low-pitched ‘peeeww’ sound. A sine wave quickly descending in pitch pretty much does it. Messing with waveforms and filters and stuff will change the sound, as will the slope of the envelope and stuff.

Snare Drums

Snare drums are trickier to make, but they’re pretty much made of just two parts. One ‘peew’ sound, higher pitched than a bass drum, is layered on top of a ‘pshht’ white noise sound, both of them quickly decaying in volume. Filtering the white noise is helpful, but generally a nice snare sound takes a million years of tweaking to make.

Clap Sounds

I hate drum machine clap sounds, but they’re interesting if you look at their waveforms. Claps are pretty much the same as snares, but they’re all stuttery, which makes them sound ‘clappy’. White noise with a stuttery volume envelope will make a pretty clap-esque sound, and it can be useful to put sounds like this in snares and stuff.

Hi hats

Hi hats are pretty much just white noise with a really quickly decaying volume envelope and some filtering. Open hi hats decay a lot slower and might need some tweaking with an envelope on the filter’s cutoff frequency.

If you try this out, the sounds you make will probably sound dumb, but that’s the entire point. A whole bunch of distortion will make them sound uber haaaardcore, so don’t worry too much about it.

A look at some cheesy drum sounds

So, to start off, I loaded up my favourite cello soundfont and recorded a single note into edison. It doesn’t really matter what note it is, since I’m going to only be using a single waveform. Looking at just the waveform of the recording, each cycle looks pretty much the same, but zooming out and

Zooming out

looking at the spectrum of the recording, I can see that the harmonics are morphing around, pretty much between two different states. I zoomed in on the waveform of each slightly different section, copied a single cycle and pasted it into a new instance of edison. After fading the beginning and end in and out to make sure they begin and end on zero-crossing points, I’m ready to drag them into Sytrus.

Once I have Sytrus open, all I need to do is drag the waveforms from the edisons into an oscillator. So now I have two oscillators making cello-ish sounds, but they’re completely static and it sounds bad. I’m going to put an LFO on the each oscillator’s volume, but have the LFOs completely out of phase. This way, when one oscillator fades out, the other one fades in, and the sound will morph between the two. Each LFO needs to have the same rate, or weird things will happen.

Drag n Drop

 I set the volume LFO on the first oscillator, and then copied the state into the second oscillator’s volume LFO. To set them 180 degrees out of phase, I just dragged the handle thing to the bottom in the second oscillator, instead of the top. Global LFOs might work better because the phase of the ‘oscillator morphing’ would be pretty much random between each note, but maybe not.

That’s pretty much it. Sending both oscillators to the same filter and putting an envelope on the volume of the filter with a slowish attack and sending the filter to the FX section with some reverb n stuff will make it sound pretty okay. I decided to mix it with some detuned+distorted s

LFO

aw waves, and it sounds freaking cool.

I have a MIDI keyboard controller, and it has 9 faders and 8 knobs that send CC stuff. Basically, these knobs can control knobs on the screen. Controlling only 17 things is kind of limiting, there are a lot more than 17 things I’d like to control. In some other blog post, I’m pretty sure I wrote about changing the MIDI channels for the knobs on the keyboard to increase the amount of things I could control, but today I wanted to take a new approach. In yet another blog post I ranted about how much I hate digital synthesizers with minimal knobbage and a bunch of pages to scroll through, but I decided to do exactly that, except with software.

Soooo, here’s what I’m trying to do in SynthMaker. All of my 17 knobs and faders will control 17 virtual knobs n stuff on the screen, but one of them will be able to scroll through different ‘pages’ and change what all the other knobs control. On page 1, the knobs control 16 oscillator parameters, on page 2 they control 16 filter parameters and whatever else. The actually knobs that are being turned are always controlling the same virtual knobs, but what those knobs are controlling can change. It’s like… instead of many light switches turning on many lamps, the switches on the lamps are turning on different lights. Actually, that’s a very bad analogy.

To figure all this stuff out, I’m just using two knobs. One controls either the pitch or the volume of a sound, depending on the position of the other knob. These virtual knobs send out numbers from 0 to 1 and those numbers are connected to other stuff. It’s pretty easy to make a switch where the numbers from knob #2 go to different places depending on knob #2. If I turn knob #1 and adjust the pitch and then turn knob #2 so that knob #1 controls the volume, the pitch stays the same, but if I turn knob #1 to adjust the volume and then turn knob #2 to control the pitch again, it automatically ‘jumps’ to where the knob really is, messing up the pitch. What should happen is when I switch back to control the pitch, the pitch only changes after I move the knob that it’s connected to, so when I’m switching pages the different parameters ‘remember’ where they were until I tweak the knob that’s currently controlling them.

I’m not entirely sure how I’m going to do this. Okay I think I did it, mostly by screwing around aimlessly. So, knob #1, or the page knob, sends out numbers from zero to one. These numbers are rounded to the nearest integer (0.5 becomes 1, blah blah blah) and this number is checked to see if it equals 1. If this number is equal to 1, the second knob doesn’t control the pitch, which isn’t what I want, darn. I’m going to end this post now, and keep wasting time.