The YouTube trailer would probably look something like this:

I hacked Angel Harp together in my spare time. Four long months! The plan was to finish by Halloween of 2011, but it took considerably longer than expected. The synthesis was completed in one week, the sound effects in another week. Standing on the shoulders of Twang, Angel Harp produces somewhat-realistic tones (like an actual harp! Complex filtering!) And it has 3+ dozen strings, for serious plucking power!

And, the graphics… Let’s talk about that.

Once the Halloween deadline became improbable, I decided to hack each feature until it was “good enough.” If any feature became an eyesore, then I’d revisit it — either for version 1.0, or a future release. The clouds were redone a couple times. I had grand plans for the harp itself, using an (awful, buggy) harp modeling tool; in a future version, you can draw your own harps, and skin them with fancy materials, I think.

How do you like those smiles? The facial expressions were sketched in 30 seconds, all of them. In my mind, the enormous half-circle grins and grimaces were very temporary. However, as friends were subjected to the ‘Harp prototypes, it became clear that people loved the primitive scribbled angels, and they wanted to keep them. Beware the things you bring to light.

I have mixed feelings about the facial expressions. Actually, I love them. They’re so happy. It’s almost a parody of top-selling app mascots; googly-eyed characters, managing diners and flying over the outback, are positively leering. Like they haven’t tasted human flesh for millennia, and then suddenly YOU show up.

I worry that prospective buyers won’t risk a couple bucks on Angel Harp, because it might look a bit hastily designed. (Even though it feels reactive and expressive, the screenshots can’t communicate that.)

Angel Harp was a conscious attempt to entertain all ages, and still deliver something unexpected and quirky. Everyone who interacts with Angel Harp loves it, but it’s a big, intimidating knot of weirdness to figure out. Is it an instrument? Why are there angels? Why do they squeal so much?

Frustrating admission: I must create apps that are more familiar to the average person (or a certain subgroup of insane musical geniuses).

Not-so-frustrating admission: I’m proud of the Devil Harp’s carousel of effects. The reversed audio, the tube amp distortion, and the fuzz box. Don some headphones, and lose yourself in the Devil Harp’s ambient spookiness. It’s nice.

Career-building achievement: I greatly expanded my OpenGL knowledge.

True tales of bits and bytes: The sound effects are stored in one continuous file, which sounds like this, and can cause hysteria:

Culinary trivia: Coffee is a wonderful drug. It powered both recording sessions. My arms flapped wildly, as I voiced every angel and demon.

Creepy outtake: Angel Harp mostly “designed itself,” but an early version of the Hell world was genuinely upsetting:

Will there be more Control Z apps in the future: I expect so! There’s one music app that I’m itching to create, but it’s a massive undertaking. Work, play, life, action, rest… I’m still finding the right balance. We shall see.

Is it better to create simple apps, which have one function: Almost certainly. I always want to go beyond the limits, and tinker with new, unprofitable things. This is my curse. If you’ve figured out how to manage this affliction, let me know.

View Angel Harp at the App Store!

The rules of Song Pact were simple: Two original compositions per month, from each of us. If you succeeded wildly, and submitted three (3!) original songs, then you could treat yourself to wine and chocolate, at your expense. But if you failed to submit a track in time, you had to walk a mile wearing a bag on your head.

Over time, we loosened the rules to permit remixes and cover songs, but only if the third Thursday was an even number. Also, songs must sound like real effort was expended, so five minutes of aimless keyboard noodling was disallowed.

The “Curious Lack of Light” tracks were birthed at the beginning. I had days to burn, relentlessly tweaking each mix until it became presentable:

Then a few things happened in slow motion: My sound design skills improved, so I could visualize sounds in my head and then construct them. I produced better mixdowns in less time. But I became risk-averse, and my compositions felt stale. I realized that quality music demands many dedicated hours; “serious time.” Yet, as my commitments ramped up, my free hours were spent elsewhere. I stopped scrutinizing ideas, and learned to roll with whatever notes emerged first, even if they were crap, or recycled from other material.

I learned how to make passable music quickly.

I learned that electronic music has several important pillars: sound design, arranging, and mixing. The latter two still feel like alien worlds to me, and I fumble a lot.

I’m quitting, for now. When arranging and mixing became crucial, Song Pact felt like an unpaid version of my day job (tight deadlines; brain-bending puzzlers; creative work under pressure). With so few hours at my disposal, the process was repetitive and frustrating. The approach felt wrong.

My paradigm needs to shift. The process of music production almost never changes: The human uses machines, and assembles the entirety of a track. What would happen if we incorporated machine learning? Or, can you draw an abstract “track” on paper, and convert it to a full composition? What if the end product wasn’t a fixed slice of audio, but had interactive dimensions? Electronic music may still be in its infancy. We haven’t even begun to have our minds blown.

And speaking of paradigm shifts, it was deeply satisfying to hit the stage last night and unleash this head-banging performance:

Only a handful of these tracks are “mine,” yet the diversity makes the set stronger. If you love electronic music, then this is a great time to exist on planet Earth; you are rich beyond belief. There’s an absolute glut of electronic music available now, in so many styles and micro-genres. The granularity is astounding. If you don’t feel like paying for it, you can still find DJ sets all over the web, for free.

You’d need many, many lifetimes to comb through all of this material. For each track I bought, downloaded and played live, I must have rejected a hundred others. It is ridiculous out there.

I hate quitting, but it’s time to move on. I survived a year of Song Pact. (My cartooning obsession lasted a year, as well. Perhaps that’s a good yardstick for a serious endeavor.)

I want to create art in ways that don’t exist yet.

And, since you made it this far, behold my Song Pact creations! Some of these show real potential, but not all of them! I’ve spared you some of the truly hideous specimens, with trumpet improvisations and that sort of thing:

Please note: This code comes with no warranty, nor support, whatsoever. None. Zip. Nada. If your talking robots become self-aware and enslave humanity, then I will not be held responsible. But if you’re in the mood for tinkering, here’s how it’s strung together:

• The sound is generating using Linear Predictive Coding (“LPC”).
• First, some Python code: The analyze_lpc.py script analyzes phonemes.dat (which is just a headerless version of phonemes.aif). Individual phonemes are separated by moments of silence, so the script splits the sound file on those. Each phoneme is converted to LPC data, using code that I ported from the rt_lpc project. I felt like I understood the mathematics 3 years ago, but I doubt I could explain it today.
• Now, in Flash: Launch the DictCompressor application, and watch the trace messages. Click the screen to open the browser window, then select your cmudict___.txt pronouncing dictionary. (You can obtain the latest CMUdict here.) Flash will convert this to a (smaller) cmudict.dat file, which is what LPCsynth.swf loads.
• LPCsynth is the application that talks. The LPCSynthHarness.sayItNow() method creates an array of LPCFrames, which are “spoken” in the sampleData() method.  This was never intended for public distribution, so the code is not exactly stellar (the talking bit should be extracted into its own class).

Is this interesting? Did your Flash Player become self-aware after hearing its own voice? Let me know!

It’s funny, originally this was intended for the controlzinc.com website. The robot voice would sing as you clicked, crooning about your mousing habits. I still can’t decide if that idea was brilliant, or terrible.

If only Mixing Secrets For The Small Studio had existed 10 years ago, my music would have been impeccable! (Well, I like to think so.) This book is a magical tome for anyone who records or produces music on a budget. It’s packed with big reveals, and explains the science behind each mixing technique. Forget the accumulated hit-or-miss wisdom of the internet; after reading this book, I found that I could produce substantially better mixes immediately. That’s amazing. (My mixes still aren’t great, but I’m working on it!)

Here are my favorite takeaways from the 20 chapters. I’m writing this to lock these concepts in my head. I’m skimming lots of material, because there’s so much valuable information packed into this book, I can’t possibly recap all of it.

Part 1: Hearing and Listening

My Rokit KRK-8 studio monitors are distorting my mixes. They’re “ported,” meaning they have an open-air port cut into the body. The manufacturer claims that this helps the air move freely. As a consequence, this creates noisy air turbulence, and muddies the bass frequencies. That’s bad. The book suggests plugging the ports with socks; the sound might become clearer! Try it, listen carefully, trust your ears…

The goal of soundproofing your studio is to tame reverberant frequencies. You don’t want an echoless space. Also, stereo is overrated, try mixing in mono. If this advice sounds heretical, consider the typical consumer’s listening environment: Computer speakers, stuffed in a bookcase, clear across the room. Car speakers, blaring inside a noisy, reflective space. Restaurant speakers, bolted to the ceiling in semi-random locations. Your mix should sound good in mono, because most people will never give it a proper stereo listening. And there will always be room reflections in any listening space, so don’t completely Nerf your room. Allow your walls to sweeten the sound.

Headphones are a special exception. The book cautions that some headphones have inadequate bass response (as do some studio monitors, of course). Music may sound oddly “clean” without the natural reflections of a room. Personally, I love modern headphones. I bought a pair of Beyerdynamic DT 770 headphones this summer. It was the single best musical purchase I’ve ever made. They have clear, deep bass response, which rattles my skull a bit. The treble is impressive, and it reaches very high into the range of human hearing. I can finally hear the full audio spectrum (unlike running my Rokit KRK-8′s through a Mackie mixer), so the headphones have been a godsend for me. I can’t imagine returning to my KRK-8′s in my oddly-shaped studio space.

The biggest takeaway from this section is learning how to use other songs as reference material. Create a workflow where you can rapidly switch between your mix, and several reference tracks, to keep your ears fresh. Listen to how your mix compares with the professional’s. The author repeats this throughout the book: It is possible to create stunning mixes in a small studio environment. Checking your work with professional mixes is crucial.

Part 2: Mix Preparation

It’s okay to use pitch correction, rhythm correction (such as Ableton Live’s time markers), recreate the bassline on a synthesizer and mix it in, or rearrange the band’s song. Oh, yes indeed. They’re paying you to produce good music, and if that means fixing their mistakes, and using smoke and mirrors, then so be it!

Consider the overall flow of the song. Each instance of the chorus could have a different intensity. Perhaps they’ll get (perceptably) louder, except for the “drop chorus” at the end: Play the chorus once, with some instruments cut out, then play another chorus with everything driving the song home.

You may need to “mult” recorded tracks, which means splitting them up into separate layers, with different processing for each layer. The rhythm guitar should subtly slip into the background when the vocalist is singing, for example. Now that I’m aware of multing, I can’t help but hear it used everywhere!

Part 3: Balance

Every layer will need a high-pass filter, probably no steeper than 18db/octave. Raise the filter frequency until the track feels like something is missing, then lower the frequency a bit. This will liberate the precious low-frequency spaces, so the kick drum and bass won’t have to fight the other instruments in a bath of bass mud.

Now it’s time to balance the levels. Mute every track. Slowly unmute the trocks, one at a time. Start with the most important section; start with the most important instrument. Adjust the track’s volume so it fits with the mix, then don’t touch it afterwards. After you’ve leveled every track, your mix should now sound pretty good.

Compression is primarily used to stabilize the balance of individual layers. Some tracks will have a steady balance without compression, and that’s okay! Leave them alone. Other tracks may need one compressor to reduce momentary spikes in amplitude; another compressor to help even out the amplitudes of individual notes; etc. Try using parallel compression (aka “New York compression”), sending a track through a heavy compressor on a second channel, which will help even out the sound yet preserve its original character.

Expanders (and gates) are typically not essential. If they’re used, put them before compressors in your effects chain. If your drum beat needs more punch, try sending the track to a parallel channel, gating it (to isolate just the attacks), and applying EQ and a little distortion before mixing it back in.

Equalization (“EQ”) is a broad topic. Before delving into it, I’ll disagree with one point up front: The book claims that any EQ plugin is about as good as any other. Most modern DAWs are equipped with plugins that provide low-pass, high-pass, shelving, peaking, and notch filters. My opinion is that some EQ plugins sound substantially better than others. I have never enjoyed the sound of Ableton Live’s EQ 8, I feel that it sucks precious high-end frequencies and detail out of anything it touches. EQ 3 is better, although it’s less flexible, and the filter curves are steeper, which is not always what you want. Shameless product placement: My new favorite is Tone2′s FilterBank3.

To EQ the mix, mute every track, then unmute and EQ them in order of importance. You’re trying to avoid frequency masking, a phenomenon where layers obscure each other. Avoid graphic EQ, and try shelving filters before using peaks. Use low “Q” values, so your filters will have wide, smooth curves.

Bass instruments: Try adding high-end brightness to the bass guitar. Try reducing the kick drum around 400hz, creating a hole for the bass guitar to punch through.

If a track’s dynamics aren’t doing a good job, apply EQ before the dynamics. (It’s not uncommon to use both pre- and post-dynamics EQ, although this is more difficult to wrap one’s head around.)

Linear EQ should generally be avoided, since it smears the clarity & definition of transients (momentary bursts of sound, such as drum attacks).

And, this chapter answered an old riddle for me: Why are we told to use EQ cuts, and avoid EQ boosts? Because when you apply EQ to a range of frequencies, that frequency range suffers phase alterations. This can rob sounds of their clarity. As you boost phase-altered frequencies, you’re emphasizing muddier, out-of-phase frequencies. So a high-shelf cut (with a gain after it) really is preferable to the “equivalent” low-shelf boost. In a multi-mic environment, EQ boosts can be catastrophic, since instruments will leak into neighboring microphones with slight delays (i.e. different phases!), and may induce bad comb filter effects. Thank you, Mike Senior. I salute you for your clear explanation.

This surprised me: distortion is often used as a mixing tool! EQ can only modify frequencies that already exist, whereas distortion filters can add brightness and sparkle to the high-end. If your distortion plugin lacks a wet-dry control, the book suggests using parallel processing, by sending the track to an effects channel, and mixing it back in.

Multiband dynamics plugins have their uses, but are “mostly just an extension of what we’ve already covered in previous chapters.” If you’re clever, there are ways to achieve multiband dynamic effects without a dedicated plugin. For instance, if your bass guitar notes have uneven sustains, try extracting the lowest frequencies using a low-pass filter, and then compressing those, and mixing them back in. Watch out for a “hollow” quality, which indicates that phase artifacts are entering the mix.

Side-chained dynamics can be useful. The book explains “ducking,” which is akin to extreme side-chained compression: When the vocalist is singing louder than a certain threshold, the guitar “ducks” a fixed amount. The brilliant thing is, if you don’t have a ducking filter, you can create one using a side-chained gate: Send the guitar through a gate (side-chained to the vocals), and then invert the output. When gate opens, the inverted signal mixes with the original guitar channel and reduces the overall amplitude. Amazing. You can get creative, too: Try adding a linear-phase high-pass filter to the gate’s output, so only the high frequencies duck out when the vocals are active. You get the idea.

Part 4: Sweetening to taste

Reverb, chorus, and delays cannot salvage a bad mix. However, they can sweeten a good mix, and make it great.

My big takeaway from the chapter on reverbs is that you can move instruments away from the foreground, into the background by sending them through a common “blend reverb.” The reverb should have about 10-20ms of predelay. Reverbs should be CPU-intensive, as the better ones use lots of simulated reflections, and this requires more CPU power.

My big takeaway from the chapter on delays is that reverbs are overrated! Delays offer the same benefits as reverbs, but occupy less “space” in the mix.

The chapter on stereo enhancement is extensive, but the main point is that stereo width is more important than stereo panning. There are a number of ways to achieve stereo width effects using plugins (even rotary speaker plugins are discussed!). If the guitarist plays the same riff repeatedly, try hard-panning two instances of the riff, left and right. Try an M&S plugin, which splits a layer into the sum and stereo difference; pan the sum to the center, and apply all the crazy effects you want to the difference. Try adding some stereo “room tone,” which can be a recording of tape hiss, background noise in a room, or the crackling of a vinyl record. Try duplicating the vocals, pitch shifting them a few cents apart, and panning them hard left & right.

Finally, there’s a chapter on the “endgame,” which is final compression, equalization, and automating individual notes & transients by hand. The big shocker for me was that a little hard clipping is acceptable. This is used in commercial music, and it can add a little sparkle to the sound. That’s wild.

Definitely, definitely reference your mix with other people’s tracks. Try sitting in silence for a minute, and hearing the song in your mind. Now play your mix; does it sound like what’s in your head?

The book recommends a myriad of useful plugins, including shareware and freeware! The pages are also sprinkled with creative advice from professional producers.

Really, if you produce music at home, you’re insane if you skip this book. It really is that enlightening. Get it! Use it! Everyone deserves to make high-quality music, and it can be done on the cheap with today’s technology.

I paid $99 for a Midines in the year 2008, and… I have still not received it. I have sent Chris Kann at least a dozen emails, and never received a single reply. In 2008, I did track him down on IRC — he mentioned that he was going through some hard times, but now it is 3 years later, and he has been completely silent.

I wondered if I offended him personally, but it turns out that many, many, many, many (etc) people have sent money, and received nothing. A critical mass of people complained to PayPal, who finally shut down his account, so he switched to using Google Cart Service this year.

If I receive a Midines, then I’ll be happy to give it a proper review! In the meantime, I urge everyone to use their own research & judgment before sending money to strangers.

Meanwhile, whether you own a Midines or not, you can make NES-style chiptunes for free. I created an Ableton Live Pack with all 5 NES waveforms: Download it here. You can also download the raw AIFFs. Here’s a silly track that demonstrates how it sounds:

NES Instruments by Too Many Moths

The 5 waveforms are: 1-bit noise (the drum beat), a 4-bit triangle wave (the bass), and pulse waves at three different widths: 50%, 25%, and 12.5%. Here’s how the pulses look, next to the noise:

The triangle wave is a bit tricky. The NES provides 4-bit volume control for the other channels, but the triangle wave always plays at maximum volume. In my Live Pack, I generated a triangle wave using Operator, downsampled it using Redux, and then filtered the high frequencies to curb the Redux artifacts. This was a subjective process, I tweaked the filter until it sounded correct.

I recommend altering the instrument envelopes, to create smooth attacks and gradual delays. Also, try using vibrato and tremelo (changing the pitch or amplitude with an LFO). This will improve the character of the instruments immensely.

If you want a more authentic sound, try limiting yourself to 2 pulse channels at a time, since that’s all the NES can support. You can also send the instruments through convolution reverb, using a small speaker impulse. This may sound more like an 1980′s television speaker. Enjoy!

Try it. Click to generate new attractors.

The attractor coefficients are still chosen randomly. But now, attractors that explode/collapse are rejected. Also, attractors that create “boring” shapes (by drawing the same pixels repeatedly) are discarded. It’s a little slow, but I’m sure the speed could be improved using Pixel Bender.

Also, here’s the source code. (Compile with Flash CS5.)

Watch: regular, or HD version. (Vimeo, 4:00)

The clips have a bias towards art and animation, because: 1). It shows well in a video, 2). This reel was created for an application to an artist’s colony. (More on that later.)

The intro & outro animations (the flying moths) were created specifically for the reel. They’re my favorite clips, yet they took less than a day to assemble. The moths were left-over from an old project (you’ll see an experimental Too Many Moths interface within the video), and I felt the convergence motion was a strong opener.

The moths were composited in Flash. They were arranged semi-randomly, then the image was “baked” onto their wings & body. I flew them backwards, and recorded their positions on each frame. Then I reset the moths and flew them forward, again tracking their flight paths.

Actually, this was done in memory, so it was very quick. My laptop needed 3 seconds to arrange, bake, & move 120 moths. Then the harness presented me with the frozen frame, so I could evaluate how well the moths covered the images. If the coverage was bad, I hit a key to build new moths from scratch.

When I found a nice arrangement, I hit another key which stepped through the animation, and rendered each frame at a ridiculously high resolution (2160×1440, to cope with scaling problems in Adobe Premiere, unfortunately). The frames were encoded as .PNG’s and uploaded to a PHP script running on localhost, which saved the 200+ images to disk. The intro took about 20 minutes to render & export. The outro, being more complicated, took over an hour.

Looking at the reel, I’m happy to present so many types of work — animation, audio programming, visual design, interactives, games, kiosks, websites. These four years have been amazing.

• Import AIFF files
• Audition and edit formant sequences in real time
• Vowel- and function-drawing tools
• Export .syx files for the Yamaha FS1R

Click to launch. Press the space bar to play the sound.

You can push your formant sequence to the Yamaha FS1R, using software such as K_Take’s FS1R Editor. Click the “Save .syx” button, and follow the instructions in K_Take’s documentation. This is a lot of fun, and breathes new life into the FS1R.

This project became much deeper than anticipated! The code includes FFT analysis (thanks Gerry Beauregard), pitch detection, a formant detection algorithm, and an AIFF parser to read AIFF files. The interface was a challenge to design and implement, and there are still many unfinished features (see the issues list).

My energy is shifting to other work, so I’ll enhance fseq-flash when time permits.

The Korg Radias synthesizer was released in 2006, discontinued after a 4-year run, and seems to have polarized all its users. Some dismiss it as “digital crap”, others hail it as a “future classic”. Who is right? Are these people snobby analog purists? Is the other group Korg’s minions, paid to drum up favor for this little silver beast?

I purchased the rack-mount model, the Radias-R, in 2006. I was relatively new to the world of “virtual analog” synthesis, and the sound of the Radias seduced me because it was so different. Think: bright, smooth, icy, crystalline, electrical.

There are many reasons to appreciate this synth. I’ll summarize a few, because I want to skip to the heart of this post: The Radias contains 64 sampled waveforms (“SynthPCM”), and they’re are a lovely, creative bunch, with lots of rich, cybernetic character. (Someone enjoyed designing these, I bet.) The comb filter is surprisingly useful and rich. Korg’s free editing software is a joy to use.

Other features are likable, despite a few flaws: The 39 (!) knobs on the front panel don’t rotate endlessly. The insert effects are impressive, but some of the best effects (the amp simulators) are restricted to monoaural output. The unison & stereo spread features are great, but you can’t target them via the modulation matrix (yet you can twist those knobs in realtime).

There’s also a snappy vocoder, which can vocode live input, or record input as “formant motion” and play it back. Unfortunately, recording is difficult, and must be triggered on the hardware. You can’t scrub, or change the formant motion speed. The Radias can store 16 formant motions, but only 8 presets are provided. I hate to admit this, but most of them are tacky… (One of the presets speaks, “Yo! Everybody in the house, say ‘Raaaaadias!’”)

How Does It Sound?

The most important feature of the Radias — which you will either love or hate — is the character of its sound. My opinion is that anyone who describes the Radias as “deep” or “warm” is wrong, just plain wrong. The Radias is the opposite of warm, fuzzy analog. It is deeply cold, in an air-sanitizing sort of way. If iciness can be sexy, this is it.

The sound has a certain sonic distance, as if the phase of one speaker is inverted (but of course, it’s not). The basic waveforms seem a bit sterile, even after summoning some analog trickery (detuning the oscillators, etc.) The sound is so smooth that it’s practically transparent. This synth sits well in the background. With its bright character, I’ve been able to drop it into many mixes, and it magically sits well with minimal equalization. But my attempts to feature the Radias as the frontman have never felt satisfactory.

How did Korg craft this particular sound? I ran some experiments to figure this out. Here is my conclusion:

In most cases, the waveforms are not synthesized in realtime. They’re pre-sampled.

The Formant provides a clue

The Radias has 9 oscillator types, including a Formant oscillator. (A formant is similar to a bandpassed sawtooth wave.) Here’s the spectrum of a formant produced by the Radias, with the narrowest formant width available. (This spectrum was analyzed in Ableton Live.) This looks acceptable, you can see the prominent peak around 800hz:

However, when we increase the formant width, instead of producing a wider hump … The peak splits into two peaks, with the same narrow widths as before:

By increasing the formant width further, the peaks move even farther apart:

Although the sound is pleasant, this is not a true formant. This behavior indicates that the Radias’ “formant” is actually composed of two waveforms. These waveforms have been pre-sampled at various resonant peaks. The Radias produces its “formant” by playing two of these samples simultaneously.

The Sawtooth misbehaves

Another Radias oscillator type is the Sawtooth wave. When played at mid-range and high frequencies, the Radias delivers the expected sound — a series of overtones marching up to infinity:

But something undesirable happens when we move down the keyboard. When playing lower notes, the high-end treble frequencies roll off:

Further down, it sounds even more muffled:

The Sawtooth waves are not being generated in realtime. They, too, are sampled. This sort of frequency roll-off implies that a sample is being played at a low frequency, and interpolated (a.k.a. “anti-aliased”) to smooth the output. I’m not sure which window function Korg chose for the interpolation, but it is the driving force behind the Radias’ character, as it filters every sound. Also note that the other fundamental waveforms (Square, Triangle and Sine) are sampled as well.

I suspect that waveform modulations (such as square wave’s pulse width modulation) may be pre-sampled as well, since they exhibit the same roll-off. By sweeping the pulse width, the Radias may be interpolating between waveforms (like using wavetables). I can’t seem to produce an audio demo to support this claim, so the wavetables may have hundreds of samples in them, too many for the human ear to discern.

However, the two effects based on frequency modulation (“Cross” and “VPM”) do modulate in realtime. They produce appropriate spectrums, even when the oscillators run at low frequencies.

The Radias features a “morphing” filter, which can move between two filter shapes (for example, a lowpass and a highpass). This is probably achieved by processing the incoming signal with two “cardinal” filters (LPF24, LPF12, HPF, or BPF), and mixing their results together. I can’t prove this, but the spaces between the cardinal points seem a little dull, as if phase artifacts are creeping into the treble, or the anti-aliasing artifacts are doubled.

These experiments helped me understand how the Radias acquired its sound. Short, single-cycle waveforms are looped and anti-aliased, resulting in smooth, hygenic timbres. Some of the longer samples (in the “SynthPCM” group) have more density, but they’re still filtered by the anti-aliasing algorithm, which makes them sound airy and hollow. The Unison and Analog Detune features can add density, but in my experience, they don’t add much thickness or richness, as it’s merely playing multiple copies of the same smoothed waveform.

Hopefully this post doesn’t read like Radias-bashing. I was more interested to discover Korg’s tricks than anything else. The Radias is suited to some very specific genres and roles, and I wouldn’t recommend it to everyone. Still, the sweet GUI editor makes this the most “fun”, accessible synth in my collection, and I keep returning to the Radias for this reason alone.

To hear some creative Radias patches, check out Jexus’ great YouTube demos: part 1, part 2.

Filtered Noise Sequencer

Drag and resize the blue blocks to change the filter frequency and width.

This sequencer is not using expensive bandpass filters. The oscillators are sine waves, which are frequency modulated with white noise. It may not sound inherently musical, but you can produce great hihats, bass thuds, and airy pitched noises.

Here’s the source code. (Requires Flash CS5 to compile.) Have fun!

A Curious Lack of Light by tooManyMoths

(click to launch)

The next version of Twang will record & save songs (this is done), and share them in some capacity (a bit more complicated). So that’s where the Twang Player comes in. The Flash version looks like it wants to be touched & strummed. I need to revisit the design and convey that Twang Player is just a music box, you can’t compose anything in the browser! …Yet.

There are performance issues on some of my machines, too. This version uses Vector.<Number> objects to handle data, but it looks like ByteArray or even BitmapData structures are the way to go.

First, do you know Lev Grossman? He’s an incredibly talented author who recently toured Portland. If you haven’t read his book The Magicians, then stop whatever you’re doing and procure a copy immediately. Without trying to spoil anything, the major college in the book is named the Brakebills College for Magical Pedagogy. Lev saw the Brakebills T-shirt that I designed for my sweetheart’s birthday present:

Long story short, my Brakebills shirts are now officially endorsed by the author! Please buy twenty of them, or thirty if the spirit moves you. Zazzle’s print quality is great. The colors are vivid, and the ink has survived many washings. Rabid fans are buying and even customizing them! I love the internet.

Lev blogged some of my other work, too (“the guy who does this has the enviably fake-sounding name of Zach Archer”). It’s true, I have an awesome pro wrestler name.

Second, my new iPhone app has landed in the App Store:

Twang is a handheld guitar. It’s easier to play than a real guitar, and is very expressive. Instead of using audio samples, Twang uses physical modelling techniques to create a more natural, dynamic sound. No two plucks are identical. Watch my grainy first video if you disbelieve.

In the next version of Twang, left-handed people will be able to switch Twang’s orientation, and serious musicians can dampen or mute strings with their fingers. And probably more! This version is already in development, and may be submitted in a week or two? Follow Control Z, Inc on Twitter if you have a ravenous thirst for updates!

Here’s something from the vaults. Aquasound was built with these requirements in mind:

• Generate sounds that aquatic animals might make
• Sounds can be “combined” somehow
• Sounds can emote

This was never used in production. I wonder if I could turn this into something? Like a paid iPhone app? ;)

Double-click the envelopes to add/remove control points. Drag lines up & down to change their curviture. The best feature is the “Combine With” dropdown, which splices the current sound with your selection. Also the “Emote” menu will play sounds with different expression.

The audio algorithm is reverse-engineered from my beloved FS1R. I generated formants in two ways (toggle the “Tonal” checkbox to hear both), the “atonal” version is closer to ring modulation than actual formants. It’s more fun if you don’t understand what the controls are doing, but if you insist: Pitch controls the overall pitch of the sound. Freq controls the center frequency of the formant (like a bandpass filter). LFOFreq and LFOWeight control a low-frequency sine wave, which can be applied to other controls via their “___LFOAmt” curves. Amp is amplitude, Width is formant width (think: width of the bandpass filter), Skirt adds distortion. Each voice has two formant generators, check “Formant Active” to enable them.

May all your bloops and crackles be happy ones!

Here’s my second batch of image reconstruction using Times New Roman characters. The algorithm is a brute-force affair: New characters are colored & positioned randomly. Any characters that make the canvas look more like the original image are saved. And that’s about it. Oh, and the font sizes start large (5120pt) and end small (10pt), so that fine details have a chance of survival.

Time lapse of the Mona Lisa reconstruction on YouTube, with silly music.

My first batch uses a different algorithm. Each canvas allocates a certain quantity of letters, and progressively mutates them, trying to mimic the original image as closely as possible. This technique is more akin to image compression. This batch is still in progress, it’s very slow. I’ll post these when they’re ready!

This took about 5 weeks to develop. Meanwhile, I’ve started another app, and I envision releasing Metal Mouth 2.0 in a few months, with more voices, and the ability to record audio.

I spent a couple hours generating strange attractors in Flash, just a simple 2D version for now. Click to play:

Click the black region to generate new polynomial coefficients and redraw. You will have to click many times to generate something interesting. That’s the nature of fractals, I’m afraid. Some coefficients are automatically thrown out if the drawing exceeds a certain size. Unfortunately, the inverse is not true: the code isn’t smart enough to trash any drawings that shrink to microscopic size.

I believe that you can stabilize any coefficients by scaling the values of each coefficient, gradually nudging them larger/smaller until the drawing is stable. I’ll try this when I get more time. I’ve been gung-ho on my first proper iPhone app, trying to finish it before Christmas! Stay tuned…

Also, here’s the source code for the strange attractor harness! Enjoy.

Today I noticed that if you right-click on yesterday’s SWF and show the redraw regions, you can see that it’s redrawing the contents of the entire stage, even though I put the scene in a scrollRect. Is it seriously rendering a scene that’s thousands of pixels wide before displaying it ?!?!?!? Oh, no. No they DIDN’T.

Today I ported the scene to the Away3D rendering engine. Here’s the result:

It’s beautiful, and provides access to low-level drawing routines, light sources, normal maps, … It was speedy at first, then slowed down considerably when I added the glowing floors. (Each glow is 16+ triangles right now, for various reasons including: I can’t render objects in my own custom order.) This makes yesterday’s version look performant, I’m reluctant to admit.

Possible next steps:

• Reduce the native 3D rendering area, see if performance improves?
• Grow beyond Flash, embrace the future and try Unity 3D?
• Dump this project, finish that iPhone game I started, make a million dollars in 2 weeks?

Click to set the focus. Use the arrow keys to move. Touch blocks to illuminate.

I’m disappointed with two things:

1). How much time is required to create a 3D engine, even a grid-based one like mine. I’ve been wrestling this project for 4+ hours every day, for a week. I feel like I must be lagging behind, but there are ten thousand things that will go wrong when developing in 3D. The paradigm is uniquely punishing, there are always edge cases where some polygons aren’t drawn correctly. This project hasn’t been a joy.

Also:

2). Flash’s native 3D is not suited for a high-performance application like this one. It would be fine if I was only spinning a few DisplayObjects in space. However, the scene above displays up to 125 Bitmaps simultaneously. (Light all 25 bulbs (3 Bitmaps each), stand in the corner facing them, and the 25-segment walls.) 125 Bitmaps would be child’s play in OpenGL. But after you light a few blocks, Flash Player chokes pretty hard.

Here’s another version that uses a BlendMode on the lightbulbs. It looks great, but its performance is even less acceptable.

Here’s an early version that uses my own 3D computations, and the Graphics API. Also it has a limited field of view, which I widened for the latest builds. The performance is surprisingly high. I abandoned my custom 3D when I reached this point; drawing lines around each cube face was expensive, so I switched to Bitmaps, and the native 3D.

The cube faces are set to width & height of 100. However, the bitmaps are higher resolution, a 200×200 region is shown. They’re being downsampled at 100×100 before they’re rendered, not by my choice.

At runtime, I get periodic warnings like these:

Warning: 3D DisplayObject will not render. Its dimensions (8238, 1628) are too large to be drawn.

What?! How is this happening? I swear that any blocks behind the camera are being removed from the Stage. (Actually, this is difficult to verify. If I shrink the scene, Flash magically applies the 3D perspective with a weird projection, and distorts everything in Lovecraftian dimensions.) Please, Adobe, tell me that you’re not rendering the scene at 8000 pixels wide, then scaling it down to my 700×400 window, frame after frame?

Also note that you, the developer, are responsible for drawing the DisplayObjects in the correct depth order (farthest to nearest), Flash doesn’t handle it automatically. This is known as “2.5D“, and it’s wildly inconvenient.

So, I’m pretty disappointed with Flash 10′s native 3D. Even with my limited 3D experience, I dislike how it renders the scene (I’m not alone in this) and the performance is obviously sub-par. This technology will not bring 3D games to the web, it cannot.

I need to decide whether to endure its shortcomings for 4 more weeks, or if I should abandon this project altogether. There are moments when you realize you’ve outgrown something you used to love, and this may be one of mine.