Jan De Cock: Laissez Faire - Laissez Passer

Jan Who?

Jan De Cock  is a Belgian artist. After exhibiting in the Tate Modern (2005), he became the first living Belgian artist to have received a solo exhibition in the New York MoMA (2008).  In 2012 he also presented his work in the Staatliche Kunsthalle Baden-Baden. Jan De Cock is the founder of the "Brussel's Art Institute",  an organization aimed at bringing students and other people in contact with the best Belgian and international contemporary works and ideas. 

Because of his appearance in the satirical TV-show "De Ideale Wereld", I was reminded of how he hit the Belgian news around february 2016 because of starting a lawsuit against some of Belgium's biggest media companies for not paying enough attention to art related topics.

Ok... so what?

In an attempt to clarify his problem with these media companies Jan De Cock wrote a 7-page manifest explaining the problems he sees with these media companies. The manifest is difficult to read, in part due to using complicated (in some cases bordering on the nonsensical) sentences. It also contains a fair share of mistakes against grammar, spelling and punctuation rules. 

If this way of writing was done on purpose, it was done in a strike of genius (and if not, it was a lucky strike) as it accomplished two things at once:

1. It ensured a lot of media coverage of an art related topic by putting a kind of "form" (a far from flawless text) over "content".
2. It caused said media to massively ridicule the man's writing skills and not paying much - if any - serious attention to the underlying message.

I don't get it...what's so brilliant about writing a wacky text and being ridiculed for it? Isn't that just dumb instead?

Well, to understand why this is brilliant, you need to read the text for its message first, rather than for its form. In what follows I've tried to condense the 7 pages into points. Note that this is my own interpretation of the manifest, which you can read for yourself in its entirety  (in Dutch!) on the web site of the "Brussel's Art Institute". 

The manifest in a few points...

Before we dive into the manifest, I'd like to make it clear that I'm in no way affiliated with Jan De Cock or the Brussel's Art Institute. I just read the text, and found it remarkable enough to write this blog entry. I didn't ask permission or get consent from the artist to explain his views as if I totally understand them, and I'm adding words never used by the author to make my point - reader beware!

The following summarizes and paraphrases the manifest - or at least what I made of it:

• Never before did the media so fiercely try to keep up the appearance that they are doing a wonderful job serving everyone with good information.
• Reality is very different. Media are being restyled all the time with one sole purpose: selling more. Content is less important than form.
• Physical media are being replaced with digital media very quickly. Digital media are volatile, whereas physical artifacts of the past are still with us today.
• Media maintain that they serve their audience a panoramic world view, but this is nothing but marketing speak. Consumers of the media believe everything that is said in those media, and what doesn't appear in the media doesn't exist.
• We, as a society, should go back to transfer the knowledge we gained through hard labor, from the moment we started back when we were still amateur-students, until the moment we mastered our field. Media are hindering us there by ignoring art related or - more generally - all difficult subjects like the plague.
• Media do not pay enough attention to artists. These artists typically are completely ignored until they become of economical value (e.g. because they gather some international attention) at which point during a few weeks everyone wants them in their superficial talk show. 
• There's a systematic discrimination and ridiculing of artists that supposedly do "not contribute economical value", and on the contrary supposedly "live of state funding" in the form of grants. But can you economically quantify the value of such an artist? Is economy the only factor in judging if something is useful? Is a narrow-minded neo-liberal world view even capable of correctly assessing the value of art? Because monetary gain needs to be maximized, everything reduces to a kind of uniform sludge, aimed at attracting the largest possible audience.
• As a consequence we live in sad times when it comes to art. Representation, fame, exposure are more important than insight and seeing past superficial form.
• Artists asking attention for art in general are dismissed as merely seeking personal attention.
• If the media continues like they do today, growing dumber in every new restyling, in an attempt to lure a bigger audience, the readers and viewers will keep degenerating with it, and their numbers will keep on decreasing until eventually no one's left. The media are committing suicide for short-term profit.
• Just like art, media should instead establish a tradition of passing on knowledge and insights, independent of politics and economy. Not merely put up a show made up by marketeers. The message is more important than the form. Media should contribute to creating a nation of critical, analytical minds capable of distinguishing banalities from real content. The current way media works makes and keeps readers/viewers dumb.

My thoughts on this...

Now independent of whether you do or don't agree with any or all of his points, I want to point out why the manifest itself, and the way the media reacted to it, actually perfectly illustrate what the artist is complaining about: 

• Exactly as he predicted, the unusual form he chose to write the manifest (using weird language and writing mistakes) caused a lot more media attention that what he would have got if it were written in perfect Dutch, thereby illustrating that indeed form prevails over content, and
• just like he predicted, the media, in an attempt to demonstrate their superiority, mostly ignored the actual contents of the manifest, instead concentrating on finding all the spelling and grammar errors and ridiculing the writer, and accusing him of attention seeking, in their most creative wordings instead.

One has to love the sweet irony of the media falling into the artist's trap with open eyes. In my eyes, the last laugh is on the artist here...

The tone of the manifest has been compared to that of Marxist pamphlets in the media, and to some extent I can agree to that characterization - things are stated in extremes and without proof, but to dismiss the message as nothing but attention seeking seems shallow to me.

One cannot deny that just about everything we see on television and in magazines tends to become dumb and dumber. Compare today's television quiz questions to those of 30 years ago and weep... Compare today's election debates with those of 20 years ago and weep even harder... Compare today's "reality shows" to... yeah.. to what? Weeping doesn't even begin to address the stupidity we're being force-fed in some of those tv-shows. Everyone is free to choose not to watch the crap, but the availability of less-than-crap to watch is rapidly decreasing and being replaced by super-crap in a fast tempo. It seems as every year adds a new level of stupidity to TV shows.

You may argue that creating abstract art, or writing obscure books that "no one" reads or difficult TV programs that "no one watches" are not likely to contribute to this Utopian nation of analytical and critical people either. I'd say it's good if they exist for those few people who wish to think and seek. You never know who they might inspire next. Preferably these works exist in a physical form that can surive the volatility of whatever new digital format happens to be the hype of the day.

Government grants for art nowadays appear to be given to those artists that lure most people... citing reasons like "public money should be useful for the widest possible public". But paradoxically these may exactly be the people who need the grants the least, since they already are popular and have ways to survive through performances or sales of their creative output.

Near the end of the 19-eighties e.g. the Belgian television had a daily tv-show program (only 5 minutes long) called "Kunstzaken" ("Art matters") that every day would talk about an exposition or a play or something cultural happening at that moment in time. Nowadays nothing comparable exists anymore on Belgian TV because 5 minutes a day of not attracting 1.000.000 viewers is an economic disaster, isn't it?

Having said all this.... maybe it's time for artists to help the audiences understand their art a bit better. Just putting an installation on the street, and expecting the holy spirit to explain it to a random passer-by by now has proved itself not to work for abstract or experimental art. If you always do what you always did you will always get what you always got. If artists somehow found a way to better explain what they do and why it matters, they might also be able to gain back some of the audience they seem to have lost.

Get up, work ahead!

So artists who might read this... blaming media is a bit easy. don't just sit there and weep over the state of matters, but use your fantastic creativity and imagination to find ways to explain what you do and why it matters. Ernest Rutherford supposedly said: "If you can't explain your research to the cleaning lady, it's not worth doing".  Postmodern descriptions of art written in difficult language in art catalogues do not strike me as the right format for this particular task (although they can happily coexist with more mundane explanations). While it may be true that some of history's geniuses were not understood during their life time, not all ignored artists automatically are geniuses. I do not believe for a second that explaining art to a wider audience somehow kills it unless it was crap to begin with.

While it may be true that too much explanation can create prejudice or preconception in an art viewer's mind, thereby possibly closing some paths to multiple or individual interpretations, people who still remember or (re)learn how to think for themselves will not be stopped by some explanation, and might even be inspired to explore the works of art more and deeper and look at them in ways they hadn't thought of before. Art is a journey and you cannot expect a single work of art to convey the whole of history you as an artist went through to arrive there, but you can provide some hooks for people to connect it to their own world.

And people of the media... admitting you have a problem is the first and hardest step to start resolving it. You too can help artists, and humanity in general, by using your creativity to explain difficult subjects to wider audiences. Crap produced by the Rupert Murdoch's of this world has done nothing to improve the state of world affairs (it has done a lot to make the Rupert Murdoch's of this world richer and more influential though) - use your power to do some good instead.

Walsh-Hadamart transformations in supercollider

Problem

Practically the whole world is using Fourier Transforms to decompose sounds into sums of sine waves. The Fourier transform then can be edited, and transformed back to the time domain to hear the effects of the editing. One question that naturally arises is if perhaps ways exist to decompose sounds as sums of something other than sine waves.

Approach

Well, as it turns out there are infinitely many ways to decompose signals into sums of other signals, and one that personally intrigues me is the Walsh-Hadamard transform which decomposes signals into sums of pulses (square waves). Walsh functions were already known and used around 1890, but it took until 1923 for Walsh to formally examine them in a mathematical context. Most of the work on Walsh functions in the context of audio processing was done in the nineteen-seventies. The fact that it is no longer popular may indicate that the results were nothing spectacular, but that shouldn't stop us from experiencing it first-hand.

Walsh functions

With the help of mathematics it can be established that we need only a subset of all possible square waves to decompose and reconstruct any signal. These waves are now known as "Walsh functions" and they can be ordered by sequency. (Note: not frequency!). Sequency is a number that corresponds the number of zero-crossings the pulse makes in the time base. Here's a representation of some Walsh functions ordered by their sequency. Sequency is not expressed in cps (cycles per second, also known as Hz), but in zps (zero-crossings per second).

If the signal is only 2 samples long, 2 Walsh functions suffice to perfectly reconstruct any such signal. If the signal is 4 samples long, 4 Walsh functions suffice to perfectly reconstruct any such signal. In general, for a signal of 2^L long, you need to combine up to 2^L Walsh functions to perfectly reconstruct any such signal. This is similar to the discrete Fourier transform where the original signal and the transformed signal both have the same length.

Walsh functions always start with +1 as their first component. For completeness I should probably mention that the even sequencies are sometimes called CAL functions, whereas the odd sequencies are called SAL functions.

WAL(2n, t) = CAL(n, t)    n = 1, 2, ...
WAL(2n-l, t) = SAL(n, t)  n = 1, 2, ...

As with sines and cosines, CAL and SAL are in essence time-shifted versions of each other.

Here's an example of decomposing a 4 sample signal into a linear combination of Walsh functions:

The signal [-1, 1, 0, -2] can be decomposed using level 2 sequencies:

[-1,1,0,-2] = -0.5*[1,1,1,1] + 0.5*[1,1,-1,-1] -1*[1,-1,-1,1] + 0*[1,-1,1,-1]

or in words:

[-1,1,0,-2]  = -0.5*sequency0 + 0.5*sequency1 - 1*sequency2 + 0*sequency3

Note that this formula shows how to convert from Walsh spectrum back to time domain using the (known) sequencies.

Ok, so how did you find that combination? Can you always do this? Is there always only one possible combination?

I'll skip the mathematics, but yes, it can always be done, and there's always exactly one possible decomposition. Some very smart people invented an efficient way to find this decomposition, and the efficient way is known as the "Fast Walsh Transform". Explaining the transform in detail, however, is way out of bounds for this article. Please refer to some external reference for the juicy details.

Looking at the example above, the fast Walsh transform of [ -1, 1, 0, -2] should give [ -0.5, 0.5, -1, 0].

And we can go back from [ -0.5, 0.5, -1, 0] to [ -1, 1, 0, -2] by using the sequencies as we did in the example above.While we did the calculations by hand in the example above, the fast Walsh transform actually has a wonderful property: it is its own inverse (except for some constant factor).

This means that if you apply the Walsh transform to a signal, you get the Walsh spectrum, and if you then apply the Walsh transform to that Walsh spectrum again, you get the original signal again (apart from some constant factor). 

What does any of this have to do with audio or supercollider?

Well, in itself nothing really.  But we can propose some experiments with this transform.

Remember that what we are really doing here is decomposing signals (think: sounds) into weighted sums of Walsh functions (think: square waves). Square waves happen to be a basic waveform used in subtractive synthesis (think analog synths!) so now it starts to sound kind of interesting doesn't it?

We already have a way to decompose any sound into a sum of square waves, and to go back from these square waves to the original signal (time domain). What if, just before we go back to the time domain, we modify the Walsh spectrum first?

What is the effect on a sound of removing all the fast square waves (= high sequencies)? (For lack of a better word, you could call it a Walsh-Low-Pass-Filter). What is the effect on a sound of removing the slow square waves (= low sequencies)? (kind of Walsh-High-Pass-Filter). What is the effect on a sound of setting all Welsh spectrum values to 0 if they happen to be smaller than some threshold? (This could be the core algorithm of some lossy data compression scheme). What is the effect on a sound of shifting the Welsh spectrum values to the left/right (a kind of Walsh-Pitch-Shifting). Can we synthesize interesting sounds by making up a new Walsh spectrum (kind of additive synthesis with square waves)? What does a walsh filter sweep sound like? What do you get if you reinterpret the Walsh spectrum as a Fourier spectrum or vice versa? Can useful/beautiful visualizations be derived from the Walsh spectrum?

It is to be expected that the auditory results will be wildly different from what we are used to hearing in transformations based on the Fourier transform (classical high-pass and low-pass filters e.g.), but that should be all the more reason to try it out, shouldn't it? Maybe you can think of really cool new applications made possible by using the Walsh transform in audio context? If so, be sure to comment :)

Walsh transform in supercollider

Before we can play with audio, we need a way to calculate the fast Walsh transform in supercollider. Since I don't know how to write UGEN's yet, I will do some calculations in the language for now.

Here's a pretty straightforward translation of this c implementation:
This code is also available on https://sccode.org/1-5bD.

If we evaluate the following code in scide:

~walsh_transform.(values:[-1,1,0,2]);

we get back the expected result:

[0.5,-0.5,0,-1]

And to check that the inverse transform works as expected:

~walsh_transform.(values:~walsh_transform.(values:[-1,1,0,2]), rescale:false);

gives the original signal:

[-1,1,0,2].

Bring on the sounds!

This walsh, this walsh, this walsh, this walsh...

My first interest is in hearing the timbres of the Walsh functions. by themselves. So let's listen to some of those. I'll take the 256 Walsh functions from level 8 (calculate them by applying the inverse walsh transform on a spectrum containing a single "1"), and concatenate 100 copies of each into a (stereo) buffer and then play the buffer.

The resulting tones pretty much sound like pulsewidth modulated pulse waves because, obviously, they *are* pulse waves.

This article has been more than long enough for now. If there's any interest in the subject I may prepare a follow-up article in which we experiment with Walsh-transform based filters.

Using the midi tuning specification (MTS) standard in supercollider

Problem?

I have a hardware synthesizer Dave Smith Instruments Rev2 ( (aka DSI Rev2)with support for MTS (=midi tuning specification) and I'm intrigued by its possibilities. How can I reprogram my synth's frequencies from supercollider? In a next post I may or may not (depending on how fast I can solve some annoying bugs :) ) explain how to download scala tunings and keyboard mappings with the Rev2 from supercollider, but for now let's concentrate on the low-level MTS messages. Most of the information in the article explain calculations, so it should also be usable for anyone trying to use MTS from any programming language.

Approach?

It will be explained how to assign random frequencies to all midi notes by composing a valid sysex message containing bulk tuning information. The explanation is somewhat math heavy, but every step is completely detailed which should help you implement something in your favorite programming language.

MTS?

Midi tuning specification is a protocol that allows users of synthesizers to assign any frequency to any midi note. You can reverse your keyboard if you like, or you can assign random frequencies to all keys, or you can explore the wonderful world of microtonality and xenharmonics. The way to assign any frequency to any midi note is by sending midi sysex messages.

Sysex?

Sysex is a hook in the midi protocol that allows for sending anything over midi. "Anything" is to be interpreted as follows: "anything that fullfills the demands made on bytes sent over MIDI, i.e. the highest bit is always 0". This constraint means that in some cases data has to be encoded in a form that is suitable to be sent over midi, and this is also the case for MTS messages. We'll go over the calculations in excruciating detail in the rest of the article.

With some exceptions, sysex messages are not standardized. Any manufacturer can implement its own sysex messages. DSI, e.g. uses sysex to perform OS updates on their instruments. Such sysex messages typically are not publicly documented.

The sysex messages used for sending tuning information, on the other  hand, are standardized. They come in a few flavors. There's a message to request an synth to send over its tuning to a computer, but the DSI instruments appear not to respond to those messages, so I guess they are not supported.  There's another message to send tuning information from the computer to a synth, and that works well. Still other messages exist to retune a few notes in real-time, but I haven't tried those yet.

MTS message structure

MTS message structure is not very complicated, but it takes a bit of explanation. The full explanation can be read in the midi tuning standard spec, but it might contain some mistakes (currently under investigation).

First of all let's explain the general message layout. 

F0 F7 7E 7F 08 01 IDX C1 C2 .. C16 TA1_0 TA2_0 TA3_0 .. TA1_127 TA2_127 TA3_127 CS F7

• F0 F7 appears at the start of any sysex message
• 7E specifies that it's a non-real-time message.
• 08 specifies that we're about to send tuning information
• 01 specifies that we're about to send a bulk request, i.e. tuning information for all midi notes
• IDX specifies a 0-based tuning index, this corresponds to the tuning index selected in DSI's global parameters menu, alternative tunings menu entry
• C1 .. C16 are 7-bit ascii codes for the characters in the name of the custom tuning
• TA1_0 TA2_0 TA3_0...TA1_127 TA2_127 TA3_127 are the actual tuning information. Tuning information for each midi note is described using three bytes, to be explained below.
• CS is a checksum, also explained below
• F7 signals the end of the sysex message

TA bytes tuning information structure

How do the three bytes TA1, TA2, TA3 encode tuning information for a midi note? Well, the first of the three bytes, TA1 is a number that says which lower midi note number is closest to the desired frequency, if the instrument would be tuned in 12-Tone equal temperament (i.e. freq  =  2^{(midinote-69)/12}*440 Hz or vice versa midinote  =  12*log₂(freq/440 Hz)) + 69, where midinote is a number 0-127, and freq is a float.

The bytes TA2 and TA3 then encode the difference in "cents" between the frequency indicated by the midi note number in TA1 and the desired frequency. The difference in cents between two frequencies f1 and f2 can be calculated as n_c  =  1200*log₂(f2/f1)), but supercollider provides some built-in functions to hide these formulas.

In the MTS message, 128 such triples TA1,TA2,TA3 are sent, one for each midi note. 

So if you wanted to retune the 50th midi note to the value of 670 Hz, you'd have to put as 50th triplet in the message the following bytes:

• the closest, lower midi number corresponding to 670 would be 76 (midi note 76, on a 12-TET tuned instrument, corresponds to a frequency of  2^(76-69)/12*440 = 659.26 Hz ). The way to calculate this 76 is by taking the floor of the formula that transforms frequency to midi note number: floor(12*log2(670/440 Hz) + 69) = 76. In hexadecimal, 76 is 4C. 

TA1_50 = 76 (or hex: 0x4C)

• the difference in cents between 659.26Hz and 670Hz would then be 1200*log₂(670/659.26) = 27.98908618457 cents. This difference in cents will always be < 100, because the reference frequency is the closest lower midi note (and the difference between two successive midi notes in 12 TET, is always exactly 100 cents).
• We have 2 bytes TA2 and TA3 to encode this number 27.98908618457. Because in midi bytes, the high bit is reserved to indicate commands instead of data, that gives us a total of 14 bits (= 0 to 2^14-1) to encode a number in the interval [0, 100] (I include 100 because the difference in cents might e.g. be 99.995). In other words, you get a resolution of 100 cents / (2^14) = 0.0061 cents. That is what we call super high precision :) 
• If you map number 27.98908618457 from the interval [0,100] to the interval [0, (2^14-1)], by means of the formula 27.98908618457 * (2^14-1)/100 ,you get 4585.4519896181, or after rounding 4585. This 4585 is the number that must be encoded in TA2 and TA3. This happens with bit masking operations:

TA2_50 = 4585 >> 7 = 35 (or hex: 0x23)

TA3_50 = 4585 & 127 = 105 (or hex: 0x69)

Checksum calculation

• The checksum is calculated as a XOR between all the bytes in the message, except for the sysex header (F0 F7), the checksum itself (obviously) and the end of sysex byte at the end (7F).
• Gotcha: at the end of the checksum calculation, be sure to "and" the result with 127 (=0x7F) to mask out the highest bit, otherwise you end up with an invalid MIDI message. This is not explicitly mentioned in the MTS standard, but it's implicitly assumed for all MIDI data bytes.

Gotchas

• Beware of limited precision of floating point operations. In some cases, it may lead to unexpected results and invalid sysex messages as a result. I encountered such problems in my supercollider implementation of the tuning message and put in a workaround.
• Be sure to check and double check that midi note numbers used as TA1 do not become negative or >127. Failing to do so will cause you hours of debugging (I should know by now!) to find out why sometimes - seemingly out of the blue - the synthesizer does unexpected and weird things when sending a new tuning (like randomly switching to a different program and changing some LFO settings) or why sending a new tuning doesn't seem to have any effect.
• The MIDI MTS spec states that sending a tuning value with TA1, TA2, TA3 = 0x7F 0x7F 0x7F should be interpreted by the synth as "do not change tuning", but it seems that DSI instruments interpret it as "tune to the highest possible frequency" instead. For this reason the code currently just puts the standard note where no note should be present.

Supercollider code

Note that the following screenshot is part of a supercollider class, part of a larger system to communicate with the DSI prophet rev2 hardware synthesizer (the code can be found here: https://github.com/shimpe/sc-prophet-rev2 ), so you'd have to change some minor syntax details to get it working as a regular function (click the image to enlarge, or - even better - copy paste it from github (which may have a more recent version): https://github.com/shimpe/sc-prophet-rev2/blob/master/Classes/ScalaCalculator.sc).

Panola: a supercollider PAttern NOtation LAnguage

Problem?

In supercollider, one of the best ways to schedule notes over time is by using the pattern system. The pattern system is very flexible but the flexibility can make it a bit hard to use. 

In a typical pattern specification, also known as a Pbind, every dimension to your sound event is independent from every other event. This is just a fancy way of saying that you need to specify volume separately from duration, separately from midinote, etc.

If you want to delete a note in the middle of your piece, you have to dig through all these keys in the pattern specification to find the right stuff to remove or add so as to make sure that all the rest still sounds as intended.

Solution?

By grouping all musical information together and extracting the required keys from that central specification, we can change information in one place and keep all the resulting keys in sync automatically.

Thinking about this, I thought back of a creation I made last year: the midi specification language (MISPEL, see https://github.com/shimpe/expremigen ). I dreamed about having this system available in supercollider. I didn't quite get there yet, but now at least a subset of Mispel is available in supercollider and that subset is known as Panola (pattern notation language), available from https://github.com/shimpe/panola 

Tutorial?

(For now this is a literal copy of the tutorial on http://sccode.org/1-5aq - feel free to go there, it has syntax coloring :) )

// Panola is a way to extract Pbind keys from a concise specification.
// This makes it easier to compose "traditional" music with Pbind, with a lot less
// headache trying to keep the different keys in sync
// It's the type of system I've missed since my day one with supercollider.

// First things first. To install Panola:

Quarks.install("https://github.com/shimpe/panola");

// Now you can get the help document by typing ctrl+D with the cursor on the word
// Panola in the next line

Panola.new("a4");

// Let's start with the "Hello world" of Panola: a simple scale.
// The numbers indicate octaves.
// You don't need to repeat octave numbers if they don't change between notes.
(
~ex = Panola.new("c4 d e f g a b c5");
~player = ~ex.asPbind.play;
)

// asPbind takes a synth name as parameter (which defaults to \default).
// So the above is equivalent to
(
~ex = Panola.new("c4 d e f g a b c5");
~player = ~ex.asPbind(\default).play;
)

// instead of calling a single "asPbind" you can also extract all information separately
// like this you have optimal flexibility in what you want to use from Panola
(
~ex = Panola.new("c4 d e f g a b c5");
~pat = Pbind(\instrument, \default, \midinote, ~ex.midinotePattern, \dur, ~ex.durationPattern, \amp, ~ex.volumePattern, \tempo, ~ex.tempoPattern, \lag, ~ex.lagPattern, \legato, ~ex.pdurPattern);
~player = ~pat.play;
)

// You can make chords using angular brackets. Only note properties of the first
// note in the chord (other than octave number and note modifier (see later)) are
// taken into account.
(
~ex = Panola.new("<c4 e> <e g> <c e g c5>");
~player = ~ex.asPbind.play;
)

// You can use modifiers on the notes:
// # for sharp, x for double sharp, - for moll, -- for double mol
(
~ex = Panola.new("c4 d- e f# gx a# b-- c5");
~player = ~ex.asPbind.play;
)


// With underscores you can indicate rhythm.
// The last used rhythm value is reused until a new one is specified:
// Here's four quarter notes (_4) followed by four eighth notes (_8).
(
~ex = Panola.new("c4_4 d e f g_8 a b c5");
~player = ~ex.asPbind.play;
)

// You can use one or more dots to extend the length of the rhythm, as in traditional notation.
(
~ex = Panola.new("c4_4. d_8 e_4 f g_16 a_4.. b_4 c5");
~player = ~ex.asPbind.play;
)

// You can also use multipliers and/or dividers to change the length.
// E.g. here we use it to create a note that lasts for three eighths
// (c4_8*3) and to create tuplets (e_8*2/3 f g). Remember that last
// duration/rhythm indication is reused until a new one is specified.
(
~ex = Panola.new("c4_8*3 d_8 e_8*2/3 f g f_16 e f e g_4 b_4 c5");
~player = ~ex.asPbind.play;
)

// Now we come to the animated property system. We can attach properties to the notes and animate them over time.
// For now two types of animation are supported: linear interpolation and fixed value.
// To indicate linear interpolation, use curly brackets {}. E.g. here we let the tempo gradually increase from 80 bpm to 160 bpm:
(
~ex = Panola.new("c4\\tempo{80} d e f g a b c5\\tempo{160}");
~player = ~ex.asPbind.play;
)

// Different properties can be combined. Here we let the volume go up until the middle of the phrase, then let it go down again,
// while tempo is rising from 80 bpm to 160 bpm.

(
~ex = Panola.new("c4\\tempo{80}\\vol{0.2} d e f g\\vol{0.9} a b c5\\tempo{160}\\vol{0.2}");
~player = ~ex.asPbind.play;
)

// If you want to use the fixed values, use square brackets instead. You can switch between fixed and animated everytime
// you specify a new property value. In the next example, tempo remains at 80 bpm until we come to note a. At that point,
// it jumps to value 100 bpm and gradually increases to 200.
(
~ex = Panola.new("c4\\tempo[80] d e f g a\\tempo{100} b c5 d e f g a b c6\\tempo{200}");
~player = ~ex.asPbind.play;
)

// Using pdur (think: played duration), we can indicate the difference between staccato and legato.
// Here we slowly evolve from very staccato to very legato:
(
~ex = Panola.new("c4_8\\pdur{0.1} d e f g a b c5 d e f g a b c6\\pdur{1}");
~player = ~ex.asPbind.play;
)

// Using lag we can modulate lag. This can be a way of creating a rubato feeling.
// Linear interpolation is not ideal for this purpose, but it's better than nothing at the moment.

(
~ex = Panola.new("a5_8\\tempo[120]\\lag{0} b c6 a5 e d c5 d e c a4 g#4\\lag{0.5} "
"a4_8 b c5 a4 e d c4 d e c a3 g#3 a b c4 d e g# a_2\\lag{0}");
~player = ~ex.asPbind.play;
)

// In addition to using predefined properties like tempo and lag, you can also use user
// defined properties, e.g. here we animate a property called "myprop".
(
~phrase = Panola.new("c d\\myprop{0.1} e f g a\\myprop{0.6}");
~pattern = ~phrase.customPropertyPattern("myprop"); // extract only myprop values as a pattern
~stream = ~pattern.asStream;
10.do({
| i |
~stream.next.postln;
});
)
// make a pbind in which the myprop appears as one of the keys, with a default value of 0 for myprop
(
~pbind = ~phrase.asPbind(\default);
~stream = ~pbind.patternpairs[13].asStream;
10.do({
| i |
~stream.next.postln;
});
)
// make a pbind in which the myprop appears as one of the keys, with a customized default value of 0.4 for myprop
// (such default values are used if no values for myprop are specified yet, e.g. in the beginning of a Panola string,
//  before any myprop is defined).
(
~pbind = ~phrase.asPbind(\default, custom_property_defaults:Dictionary.newFrom(["myprop", 0.4]));
~stream = ~pbind.patternpairs[13].asStream;
10.do({
| i |
~stream.next.postln;
});
)
// make pbind in which only the standard panola keys are included
(
~pbind = ~phrase.asPbind(\default, include_custom_properties:false);
~pbind.patternpairs.postln;
)

// These custom properties can be e.g. used to drive synth arguments
// The 303 synth used below is reused from https://sccode.org/1-4Wy
// which in turn is based on code from Lance J. Putnam
(
s.waitForBoot({
var line;

SynthDef (\sc303 , {  arg  out=0, freq=440, wave=0, ctf=100, res=0.2,
sus=0, dec=1.0, env=1000, gate=1, vol=0.1;
var  filEnv, volEnv, waves;
volEnv =  EnvGen .ar( Env .new([10e-10, 1, 1, 10e-10], [0.01, sus, dec],  'exp' ), gate, doneAction:2);
filEnv =  EnvGen .ar( Env .new([10e-10, 1, 10e-10], [0.01, dec],  'exp' ), gate);
waves = [ Saw .ar(freq, volEnv),  Pulse .ar(freq, 0.5, volEnv)];
Out .ar(out,  RLPF .ar(  Select .ar(wave, waves), ctf + (filEnv * env), res).dup * vol);
}).add;

s.sync;

line = Panola.new(
"a2_16\\wave[0]\\vol{0.05}\\tempo{120}\\res{0.2}\\sus{0}\\env{1000}\\ctf{100} a a a1 a2 a a3 a2 a a a1 a2 a3 a2 b- g\\res{0.05}"
"a2_16\\wave[0] a a a1 a2 a a3\\sus{0.2} a2 a\\ctf{3000} a a1 a2 a3 a2 b- g\\res{0.2}"
"a2_16\\wave[0] a a a1 a2 a a3 a2 a a a1 a2 a3 a2 b- g\\res{0.01}\\sus{0}\\env{10000}\\ctf{10}"
);
~player = line.asPbind(\sc303).play;
});
)

Baking sound in supercollider

Problem?

I want to synthesize gorgeous pad sounds in supercollider. This is possible using additive and/or subtractive synthesis, but it takes a lot of CPU power.

Approach

't Is the season to be bakin' fa-la-la-la-laaaaa la-la-la-laaaaa. How about we pre-render the sound into a wavetable and then loop the generated wave table? This moves most of the work to startup time. In the blender 3d program, pre-rendering heavy calculations is known as "baking".

But... won't we lose real-tome creative possibilities then? After all, you cannot modify the partials in a prerendered wavetable anymore?

Well... yes, you will lose some creative possibilities, but we can easily reintroduce some by adding amplitude envelopes, filters and filter envelopes and effects like reverb or phasing and flanging.

PadSynth

One of the more exciting pad generation algorithms no doubt is Paul Nasca Octavian's PadSynth algorithm. It is explained in detail here. I will use it to demonstrate the approach.

In essence it takes as input a basic sound, specified as a set of partials with associated relative volumes. 

It then enriches that spectrum by replacing each of the partials with a fuzzy region containing smeared out partials. This is the equivalent to adding many slightly detuned copies of the original signal, which is how sounds can start to sound a lot warmer. Since human hearing is less sensitive to small frequency differences for high frequencies than for low frequencies, the spreading of partials is made wider in the higher frequency bands than in the lower frequency bands. PadSynth also sets the phases of each of the detuned copies of the signal to random values which helps in reducing undesired comb filtering effects.

PadSynth in supercollider

Supercollider does not come with an implementation of PadSynth, but it is not too difficult to convert the pseudo-code from the detailed explanation into working code. A post by Donald Craig on the supercollider mailing list provides an excellent starting point.

I modified the algorithm a bit as follows: I calculate two different wavetables per octave over 12 octaves (this is probably a bit overkill - feel free to change it). This is an intensive calculation and takes quite a lot of memory. You can certainly reduce this calculation time and required space by reducing how many octaves you want to use, by reducing the number buffers per octave and by reducing the size of the pregenerated wave table (I generate wavetables of around 5 seconds long each).

While playing, the pitch can be chosen by looping over the generated wave tables with a different speed. By varying the speed on-the-fly one can also implement pitch bending or glissando's. All of this is illustrated in the sample code below, in which I will be using supercollider's excellent pattern system to drive a cello-like PadSynth pad. The code can also be found on sccode.org.

Driving the synth from supercollider patterns

Because the synth loops over one of many pre-generated buffers, we'll have to inform the synth about which buffer to use for a given note. For this reason you'll see some calculations being done in the Pbind and Pmono in the sample code. If you decide to generate less buffers per octave, you will also have to adjust these calculations.

More in detail: the parameter \mynote contains a midi note number. This is the note we will want to hear. From that parameter, another parameter \myreference is calculated. This contains another midi note, namely the closest midi note number that was used as a reference to pre-render a wave table: I generate 2 wavetables per octave and each of these wavetables have a reference midi note number. This information is needed in the synth to determine how fast to loop over the buffer. \referencefreq contains the same information but expressed as a frequency in Hz instead of as a midi note number. Remember that I generated two wavetables per octave, for 8 octaves. The parameter \buffer contains the buffer number to use. The buffer being used is the one that was generated with a reference midi note closest to the desired midi note. This information is needed in the synth to know which buffer to loop over. 

The full code is a bit long to paste into this article, so please head over to sccode.org to get it!

If sccode.org is down, there's still github as a backup :) 

Audio transitions in supercollider

Problem?

In the domain of images and videos, programs like powerpoint and video editors provide a wide range of "transitions". If you think of powerpoint, e.g., you can switch from one slide to the next directly ("cut"), or you can gradually fade out the first one and fade-in the second one, or you can push in, push out, wipe according to a shape, split, reveal, random bars, shape, uncover, cover, flash, morph, ... too many to name.

In the audio domain, however, the options seem quite limited. I'm only aware of direct cuts and cross-fading as audio effects that are offered by default. Can we think of other audio transitions as well?

Approach

In this blog post I will propose a few other audio transitions, as a proof of concept. The code can be found on http://sccode.org/1-57H.

Setting up supercollider

To illustrate the transitions, let's first create two sounds that we want to transition between.

\sound1 is a sine wave with randomly varying frequency (frequency changes 4x per second).

\sound2 is a sawtooth with varying duty cycle. Both sounds are different enough that we can hear the transitions taking place.

I let the 2 sound producing synths output their audio to audio busses.

Next to the sound producing synths I have a bunch of transition synths that calculate the transition between \sound1 and \sound2 as a kind of audio effect. Each transition synth takes a pos argument. Setting it to -1 corresponds to only listening to \sound1, whereas setting it +1 corresponds to only listening to \sound2. For values between -1 and 1 the transition is taking place.

For demo purposes, the pos argument in the transition synths is ignored and instead driven from a fancy counter made specified by an envelope (Env). This fancy counter ensures that supercollider first plays \sound1 for 5 seconds, then gradually transitions to \sound2  during 5 seconds, then keeps playing \sound2 for 5 seconds, then transitions back to \sound1 during 5 seconds and finally plays \sound1 again for 5 seconds.

Groups are needed to ensure that the transition synths are executed on the server after the sound producing synths. Otherwise we wouldn't get audio output. (After all, it's impossible to fiddle with a sound that isn't calculated yet.). 

In the fork section we will instantiate all effect synths one by one so we can hear all of them in succession. In the first screenshot only the "direct cut" is added. This is a transition that basically does nothing: it just stops the first sound and starts the second one in the place.

Familiar effects

cut

A cut (in audio this is sometimes called a "butt splice") is where you stop one sound and start another sound. This basically means: "no effect". In supercollider this can be implemented by stopping one sound and starting another one. This is already illustrated in the screenshot above.

crossfade or xfade

A crossfade happens when you graduallly lower the volume of one signal while gradually increasing the volume of a second signal. For a while both signals sound simultaneously. This is supported in virtually every digital audio workstation (DAW) in existence. Often one can choose whether to change the volume keeping a "constant gain" (linear curve), "constant power" (square root curve?) or even other types like "exponential" (using logarithmic curves).

In the following screenshot I indicated what was changed to add a new effect. In the next effects I will only show the new effects synths. You can easily update the fork section yourself.

So for cross fading with equal power you could do something more like

Seldom explored effects

Do we have to stop at direct cut and crossfade? Of course not... let's try something else. Note: these techniques may actually be well established, it's just that I've never come across them "in the wild". But then, also not many people use transitions in powerpoint and video editing because if you overdo them, they tend to lower the overall end-user experience.

HPF curtain

In this transition, we increase the filter frequency of a high pass filter (HPF) for the signal that is to disappear, while we decrease the filter frequency of a high pass filter for the signal that is to appear. The implementation of this transition is not perfect as it has some audible artifacts on positions close to -1 and +1. It might help to use a high pass filter with steeper roll-off, or to combine this technique with some amplitude control.

LPF curtain

Very similar to the previous transition, we decrease the cut-off frequency of a low pass filter (LPF) on the signal that must disappear, and crank up the cut-off frequency of a low pass filter on the signal that must appear instead.

Wash out

The sound that is to disappear drowns in reverb whereas he other sound emerges from the reverb.

Pixelate

 The sounds become coarser and then more crisp again. This fragment features some complication involving Select.ar because LFPulse does not behave mathematically correct for duty cycle width=0 and width=1 (it generates audible spikes, which ruin its function as an audio stream selector), so near the duty cycle = 1 and 0, I had to select pure versions of the audio.

Push out

The new sound shifts in from the left side and pushes out the old sound to the right.

Can you think of other transitions?

Scoring a movie with supercollider, blender, python and osc - part II: keyframing and animation curves

Problem?

In the previous part of this tutorial, I described a way to insert markers on blender's timeline, and to use blender's python scripting abilities to convert these markers into OSC messages which then can be interpreted in supercollider to perform commands (e.g. start and stop patterns or sound effects). This approach works very well for sending discrete commands: things like start this, stop that. If you haven't read that part yet, please go read it as you will need to understand the concepts explained there to follow part II.

In this blog post, I want to show how one can use blender's powerful keyframing system with animation curves to continuously update parameters in supercollider sketches. This technique adds extra possibilities for driving supercollider from blender.

Approach?

In the blender scene, I will add custom properties, and animate those via keyframes and animation curves. I will extend the python script I developed in part I to send the parameter value updates to supercollider.

What is a keyframe?

A keyframe represents the value (the state) of one or more parameters at a moment in time. Suppose you have an X-position of an object in a 3d scene. At time frame=0 you can set X=0 and insert a keyframe on frame=0. At time frame=100, you can set X=10 and insert a new keyframe on frame=100. Now for frames between 0 and 100, the value of x can smoothly vary between 0 and 10, that is, the values of x can be interpolated between the keyframes (you can also ask the system for an abrupt change instead of interpolation). The interpolation can be linear, or can include "easing".

Adding custom properties

Blender is extremely keyframeable. Literally every parameter you see in the UI can be keyframed. Blender is also very extensible: users can add new parameters ("custom properties") in the UI. And those new parameters can be keyframed. 

To add a custom property, first switch to blender's default view

Then, on the right hand side, go to the scene tab, and in the custom properties section, click the Add button.

As we did in part I of the tutorial, we will again use a naming convention for the properties so that our python script knows what to send to supercollider from the property name. The proposed naming convention is that a property named SC_something42 wil be sent to supercollider as osc command /something42 with as argument the property value. We will make the python script so that only changes in values are sent. Values that don't change from one frame to the next will not be resent to supercollider. The intention is to avoid creating an avalanche of useless OSC messages.

After clicking the add button, we can set up our property using the naming convention. Suppose I want to automate a frequency in some synth. A possible custom propertyname could be SC_animfreq. This would trigger sending an OSC message /animfreq to supercollider with the value of animfreq as argument. I've set the default value to 440, and the min and max to 20 and 20000 respectively.

Now on the blender timeline, navigate the playhead to - say - frame number 10 (the easiest way is to type 10 in the current frame edit box), set the value of customproperty animfreq to - say - 220. Hover your mouse cursor over the value 220 of animfreq and press the letter "i" (for "i"nsert keyframe). The edit box with value 220 should change color to indicate that a keyframe has been inserted for the custom property.

Next, move the playhead to - say - frame 100, change the value of the custom property to 880, hover the mouse cursor over the 880 and "i"nsert another keyframe.

Now as you drag the playhead between frames 10 and 100, watch the custom property value box. You should see the number change gradually. This is the linear interpolation that happens between keyframes by default.

If we want to alter this behavior of linear interpolation, we need to switch to the "graph editor". A full explanation of everything that an be done in the graph editor falls outside the scope of this blog post (i.e. you'll need to find some blender tutorials about the graph editor on youtube if you want to dive deeper - let's just say that you can do some funky stuff).

After some zooming in x and y direction you can see the linear interpolation drawn as a curve.

By pressing "T" while hovering the mouse cursor over the graph area you can change the interpolation type. Since we're animating a frequency, it makes sense to set the interpolation type to exponential. Be sure to look for tutorials on F-curve modifiers to see some fancy stuff that is possible with the curves. For the purpose of this tutorial we won't go any deeper.

So far we have added an animated custom property. Now let's tell blender to send it to supercollider in the form of OSC messages. To do so I will extend the python script developed in part I of the tutorial. The additions with respect to part I are marked in the figure.

The OSC messages can be received in supercollider in exactly the same way as in part I. The received values then e.g. can be used to set the freq argument in a synth (or the density in some cloud of sound grains, or whatever you feel like doing).

A blender file and supercollider file with a simple demonstration can be downloaded from google drive.