Last week’s work really set me up for success this week. I managed to record a dozen fretless bass phrases this week. I recorded one phrase for TriStar, 737, DC-8, and 707. I recorded two phrases for A300, 727, DC-10, and DC-9. Accordingly I only have five fretless bass phrases to record for next week. That being said, rather than get a head start on pedal steel guitar recordings, I may add more fretless bass phrases, or re-record some of the phrases I’ve already recorded in order to have more exciting bass parts.
Again, in the interest of having some visual material to share, here’s the string arrangement for DC-8. In the first six measures you can see arpeggiations of a progression in G major: B7, Em7, CM7, Em7, D7, to GM7. The final seven measures shows a static section where the upper voices slowly arpeggiate a D# diminished chord while the cello stays on an E pedal.
It has been a busy month for me, so I’m afraid this experiment is not as challenging as it could be. I used the Organelle patch Constant Gardener to process sound from my lap steel guitar. This patch uses a phase vocoder, which allows the user to control speed and pitch of audio independently from each other. While I won’t go into great detail about what a phase vocoder is and how it works, it uses a fast Fourier transform algorithm to analyze an audio signal and to reinterpret it as time-frequency representation.
This process is dependent upon the Fourier analysis. The idea behind Fourier analysis is that complex functions can be represented as a sum of simpler functions. In audio this idea is used to separate complex audio signals into its constituent sine wave components. This idea is central to the concept of additive synthesis, which is based upon the idea that any sound, no matter how complex, can be represented by a number of sine wave elements that can be summed together. When we convert an audio signal to a time-frequency representation we get a three-dimensional analysis of the sound where one dimension is time, one dimension is frequency, and the third dimension is amplitude.
Not only can we use this data to resynthesize a sound, but in doing so, we can treat time and frequency separately. That is we can slow a sound down without making the pitch go lower. Likewise, we could raise the pitch of a sound without making the sound wave shorter.
Back to Constant Gardener. This patch uses knob 1 to control the speed (or time element) of the re-synthesis. Knob 2 controls the pitch of the resynthesis. The third knob controls the balance between the dry audio input to the Organelle with the processed (resynthesized) sound. The final knob controls how much reverb is added to the sound. The aux button (or foot pedal) is used to turn the phase vocoder resynthesis on or off.
The phase vocoder part of the algorithm is sufficiently difficult such that I won’t attempt to go through it here, rather I will go through the reverb portion of the patch. As stated previously, knob four controls the balance between the dry (non-reverberated) and the reverberated sound. Thus value is then sent to the screen as a percentage, and is also sent to the variable reverb-amt using a number from 0 to 1 inclusive.
When the value of reverb-amt is recieved, it is sent to a subroutine called cg-dw. I’m not sure why the author of the patch used that name (perhaps cg stands for constant gardener), but this subroutine basically splits the signal in two, and modifies the value the will be returned out of the left outlet to be the inverse of the value of the right outlet (that is 1 – the reverb amount). Both values are passed through a low pass filter with cutoff frequency of 5 Hz, presumably to smooth out the signal.
The object lop~ 10000 receives its input from a chain that can be traced back to the input of the dry audio coming from the Organelle’s audio input. This object is a low pass filter, which means that the frequencies below the cutoff frequency, in this case 10,000 Hz, to pass through the filter, which in return attenuates the frequencies above the cutoff frequency. More specifically, lopis a one-pole, which means that the amount of attenuation is 6 dB per octave. A reduction of 6 dB effectively is half the power of the original. Thus, if the cutoff frequency of a low pass filter is set to 100 Hz, the power at 200 Hz (doubling a frequency raises the pitch an octave) is half of what it would normally be, and at 400 Hz, the power would be a quarter of what it would normally be.
In analog synthesis a two pole (12 dB / octave reduction) or a four pole (24 dB / octave) filter would be considered more desirable. Thus, a one pole filter can be thought of as a fairly gentle filter. This low pass filter is put in the signal chain to reduce the high frequency content to avoid aliasing. Aliasing is the creation of artifacts when a signal is not sampled frequently enough to faithfully represent a signal. Human beings can hear up to 20,000 Hz, but audio demands at least one positive value and one negative value to represent a sound wave. Thus, CD quality sound uses 44,100 samples per second. The Nyquist frequency, the frequency at which aliasing starts is half the sample rate. In the case of CD quality audio, that would be 22,050 Hz. Thus, our low pass filter reduces these frequencies by more than half.
The signal is then passed to the object hip~ 50. This object is a one-pole high pass filter. This type of filter attenuates the frequencies below the cutoff frequency (in this case 50 Hz). Human hearing goes down to about 20 Hz. Thus, the energy at this frequency would be attenuated by more than half. This filter is inserted into the chain to reduce thumps and low frequency noise.
Finally we get to the reverb subroutine itself. The object that does most of the heavy lifting in this subroutine is rev~ 100 89 3000 20. This is a stereo input, four output reverb unit. Accordingly the first two inlets would be the left and right input. The other four inlets are covered by creation arguments (100 89 3000 20). These four values correspond to: output value, liveness, crossover frequency, and high frequency dampening. The output value is expressed in decibels. When expressed in this manner we can think of a change of 10 dB as doubling or halving the volume of a sound. We often consider the threshold of pain (audio so loud that it is physically painful to us) as starting around 120 dB. Thus, 100 dB, while considered to be loud, is 1/4 as loud as the threshold of pain. The liveness setting is really a feedback level (how much of the reverberated sound is fed back through the algorithm). A setting of 100 would yield reverb that would go on forever, while the setting 80 would give us short reverb. Accordingly, 89 gives us a moderate amount of reverb.
The last two values, cross over frequency and high frequency dampening work somewhat like a low pass filter. In the acoustic world low frequencies reverberate very effectively, while high frequencies tend to be absorbed by the environment. That is why a highly reverberant space like a cave or a cathedral has a dark sound to its reverb. In order to model this phenomenon, most reverb algorithms have an ability to attenuate high frequencies built into them. In this case 3,000 Hz is the frequency at which dampening begins. Here dampening is expressed as a percentage. Thus, a dampening of 0 would mean no dampening occurs, while 100 would mean that all of the frequencies about the crossover frequency are dampened. Accordingly, 20 seems like a moderate value. The outlets from pd reverbare then multiplied by the right outlet of cg-dw, applying the amount of reverb desired, and sent to the right and left outputs using throw~ outL and throw~ outR respectively.
For the EYESY I used the patch Mirror Grid Inverse – Trails. The EYESY’s five knobs are used to control: line width, line spacing, trails fade, foreground color, and background color. EYESY programming is accomplished in the language Python, and utilizes subroutines included in a library designed for creating video games called pygame.
An EYESY program is typically in four parts. The first part is where Python libraries are imported (pygame must be imported in any EYESY program). This particular program imports: os, pygame, math, and time. The second part is a setup function. This program uses the code def setup(screen, etc): pass to accomplish this. The third part is a draw function, which will be executed once for every frame of video. Accordingly, while this is where most of the magic happens, it should be written to be as lean as possible in order to run smoothly. Finally, the output should be routed to the screen.
In terms of the performance, I occasionally used knob 2 to change the pitch. I left reverb at 100%, and mix around 50% for the duration of the improvisation. I could have used the keyboard of the Organelle to play specific pitched versions of the audio input. Next month I hope to tackle additive synthesis, and perhaps use a webcam with the EYESY. Given that I’ve given a basic explanation of the first two parts of an EYESY program, in future months I hope to go through EYESY programs in greater detail.
This was a productive week. I finished with the bass harmonica, recording phrases for the B sections of 727, DC-10, & DC-9. Thus, I was able to get a head start on recording fretless bass recordings. I was able to record two fretless phrases for each of the following movements: TriStar, 737, DC-8, 707, and 747. I was also productive outside of the recording project, finishing a grant report, a conference presentation, some sound design work, and a conference presentation proposal.
In the interest of having some visual content on the entry I’ve included the score for the string parts for DC-10. In the B section, the beginning of the example, the strings only use the notes G, B, C, D, & F#. In the A section, the strings reduce to only using G & E, ending on harmonics. This will be one of two movements that use tremolo in the string parts.
Week two has not been as productive as I had hoped. Part of that is due to labor day, part is due to a trip to Boston on Tuesday, and part of it is due to a cursed plumbing job that would never end. Finally, on Wednesday, the external hard drive I use for recording started acting quirky, so I went out and bought a new drive, and spent many hours transferring about a terabyte of data to the new drive. I am glad to report that that process went well, and all my data is safe.
Before going through my productivity for the week, a bit of background on the project. The album will consist of nine pieces. Each piece is in ternary form (ABA). The lead guitar the drum machines and the automated synthesizers pretty much play throughout each piece. This material is already fairly thick and robust. For the additional instruments I am recording I plan to have them play one phrase for each section. Since I completed work on the theremin recordings last week that means there will be 27 theremin phrases on the album (3 x 9 = 27).
The reason I was able to complete the theremin recordings last week is I went into my sabbatical having already recorded many theremin and bass harmonica recordings. To complete my bass harmonica recordings I only needed to record seven more phrases. That being said, I only got four phrases recorded this week, so next week will be a light week for bass harmonica, and I will likely get a head start on recording fretless bass next week.
The four recordings I made this week were all for B sections, specifically, for: TriStar, 737, DC-8, and 707. I feel that bass harmonica is an integral part of my sound as Darth Presley. I love the sound of a bass harmonica. In fact, in terms of satisfaction per dollar, my bass harmonica is one of my favorite instruments. That being said, bass harmonicas manufactured in most countries are actually quite expensive. There are, however a few Chinese manufacturers that make budget instruments. The instrument I have is a Swan bass harmonica, which can be purchased for under $200.
Since this is a light week, I’ll comment a bit on the string arrangements. As I stated last week, I finished the string arrangements. Over the long weekend I formatted them on paper, and created parts for each of the four instruments. The recording session is booked for late October. The musicians will be listening to a click through headphones while recording, so I should be able to just drop the recordings into place once they are edited and mixed.
Each of the string arrangements covers the transition from the B section to the return of the A section. For each movement, the number of pitches used in the B section is greater to or equal to the number of pitches used in the A sections. To put it another way, the A sections have a limited number of pitches (as little as one, and no more than six), while the B sections tend to have a much greater variety of pitches (at least three, and as many as nine).
In the interest of leaving the reader with an image to look at for the week, I’ll leave you with the score for the string arrangement of TriStar. All of the string arrangements for Rotate tend to have a similar profile to this movement, that is the pitch tends to go up, and tends to crescendo into the arrival of the final section. In this case, that section uses only the note G.
The first week of my sabbatical went better than I had expected. I managed to record all of the theremin parts I had hoped to, leaving me a week ahead of schedule. All in all I recorded 13 phrases:
1 phrase for Tristar 1 phrase for 737 1 phrase for A300 3 phrases for DC-8 1 phrase for 727 1 phrase for 707 1 phrase for DC-10 2 phrases for DC-9 2 phrases for 747
The majority of these phrases were recorded using a Moog Etherwave Plus to control a Behringer System 55, which is a clone of the Moog modular synthesizer from the early 1970s. A few of the phrases were recorded using the Etherwave Plus to control a small Eurorack synthesizer centered on the 2HP Vowelformant synthesizer and a the Calsynth Monsoongranular synthesizer. One of the tracks was recorded using just the Etherwave Plus, yielding a traditional theremin sound. Below you can see a patch on the system 55 using four oscillators, a low pass filter, and a VCA.
Due to the new tracks, I’ve re-released recordings of TriStar, 737,and 707. As mentioned in my previous post, I had about a month of data loss. I managed to recover the string arrangements I wrote for TriStar, A300, DC-8, & DC-10, and wrote string arrangements for Rotate 737, 727, 707, DC-9. & 747. Thus, all the arrangements are completed. They only need to be formated for printing so I can send them out to the string quartet that will be recording them. Also relevant to the discussion is my left ear is healing nicely. It may not be at 100% yet, but it is getting there.
Given that I am one week ahead in my schedule, I’m revising my schedule thusly . . .
On Monday I officially start my sabbatical. My project is to complete my (as Darth Presley) forthcoming album Rotate. I have already released preliminary versions of: “TriStar,” “737,” “A300,” “707,” and “DC-10.” In the interest of productivity, I’ve created a weekly schedule, and plan to give weekly updates on my work. A semester is approximately 15 weeks long, so for each week I’ve noted what instrument I plan on recording. All but the strings are instruments I will be recording myself. Since I don’t play 15 different instruments, I have assigned the instruments that I want to be the core sound of the album to two weeks, and the lesser “color” instruments each only get one weeks worth of attention. For the strings, I plan on hiring a string quartet in Providence to play some string parts I am writing.
I may also choose to sneak in some recording on a piano as well.
Ultimately I reserved the right to alter the schedule. I have some temporary frustrations with the project as my laptop died, and I’ve lost about a month’s worth of data. Fortunately all my recordings are safe as I do that work on an external drive. However, I’m waiting on a replacement laptop. Likewise, I’ve developed an ear infection, so currently my left ear is missing it’s to 3-4k of hearing. I was told it could take two months to fully heal. In the meantime I think I can come up with work arounds for fine tuning the stereo mixes.
As was the case with Experiment 4, this month’s experiment has been a big step forward, even if the music may not seem to indicate an advance. This is due to the fact that this is the first significant Organelle patch that I’ve created, and that I’ve figured out how to connect the EYESY to wifi, allowing me to transfer files to and from the unit. The patch I created, 2opFM, is based upon Node FM by m.wisniowski.
As the name of the patch suggests it is a two operator FM synthesizer. For those who are unfamiliar with FM synthesis, one of the two operators is a carrier, which is an audible oscillator. The other operator is an oscillator that modulates the carrier. In its simplest form, we can think of FM synthesis as being like vibrato. One oscillator makes sound, while the other oscillator changes the pitch of the carrier. We can think of two basic controls that affect the nature of the vibrato, the speed (frequency) of the modulating oscillator, and the depth of modulation, that is how far away from the fundamental frequency does the pitch move.
What makes FM synthesis different from basic vibrato is that typically the modulating oscillator is operating at a frequency within the audio range, often at a frequency that is higher in pitch than that of the carrier. Thus, when operating at such a high frequency, the modulator doesn’t change the pitch of the carrier. Rather it functions to shape the waveform of the carrier. FM Synthesis is most often accomplished using sine waves. Accordingly, as one applies modulation we start to add harmonics, yielding more complex timbres.
Going into detail about how FM synthesis shapes waveforms is well beyond the scope of this entry. However, it is worth noting that my application of FM synthesis here is very rudimentary. The four parameters controlled by the Organelle’s knobs are: transposition level, FM Ratio, release, and FM Mod index. Transposition is in semitones, allowing the user to transpose the instrument up to two octaves in either direction (up or down). FM Ratio only allows for integer values between 1 and 32. The fact that only integers are used here means that the result harmonic spectra will all conform to the harmonic series. Release refers to how long it will take a note to go from its operating volume to 0 after a note ends. The FM Mod index is how much modulation is applied to the carrier, with a higher index resulting in more harmonics added to a given waveform. I put this parameter on the leftmost knob, so that when I control the organelle via my wind controller, a WARBL, increases in air pressure will result in richer waveforms.
As can be seen in the screenshot of main.pd above, 2opFM is an eight voice synthesizer. Eight voice means that eight different notes can be played simultaneously when using a keyboard instrument. Each individual note is passed to the subroutine voice. Determining the frequency of the modulator involves three different parameters: the MIDI note number of the incoming pitch, the transposition level, and the FM ratio. We see the MIDI note number coming out of the left outlet of upack 0 0 on the right hand side, just below r notesIN-$0. For the carrier, we simply add this number to the transposition level, and then convert it to a frequency by using the object mtof, which stands for MIDI to frequency. This value is then, eventually passed to the carrier oscilator. For the modulator, we have to add the MIDI note number to the transposition level, convert it to a frequency, again using mtof, and multiplying this value by the FM Ratio.
In order to understand the flow of this algorithm, we first have to understand the difference between control signals and audio signals. Audio signals are exactly what they sound like. In order to have high quality sound audio signals are calculated at the sample rate, which is 44,100 samples per section by default. In Pure Data, calculations that happen at this rate are indicated by a tilde following the object name. In order to save on processor time, calculations that don’t have to happen so quickly are control signals, which are denoted by a lack of a tilde after the object name. These calculations typically happen once every 64 samples, so approximately 689 times per second.
Knowing the difference between audio signals and control signals, we can now introduce two other objects. The object sig~ converts a control signal into an audio signal. Likewise, i converts a float (decimal value) to an integer, by truncating the value towards zero, that is positive values are rounded down, while negative values are rounded up.
Keep in mind the loadbang portion of voice.pd is used to initialize the values. Here we see the transposition level set at zero, the FM ratio set to one, and the FM Mod index set to 1000. The values from the knobs come in as floats between 0 and 1 inclusive. Thus, typically in order to get useable values we have to perform mathematical operations to rescale them. Under knob 1, we see the transposition level multiplied by 48, and then we subtract 24 from that value. This will yield a value between -24 (two octaves down) and +24 (two octaves up). Under knob 2, we see the FM ratio multiplied by 31, and then we add one to this value, resulting in a value between 1 and 32 inclusive. Both of these values are converted to integers using i.
The scaled value of knob 1 (transposition level) is then added to the MIDI note number, and then turned into a frequency using mtof and converted to an audio signal using sig~. This is now the fundamental frequency of carrier oscillator. In order to get the frequency of the modulator, we need to multiply this value to the scaled value of knob 2. This frequency is then fed to the leftmost inlet of the modulating oscillator (osc~). The output of the modulating oscillator is then multiplied by the scaled output of knob 4 (0 to 12,000 inclusive), which is then in turn multiplied by the fundamental frequency of the carrier oscillator before being fed to the leftmost inlet of the carrier oscillator (osc~). While there is more going on in this subroutine, this explains how the core elements of FM synthesis are accomplished in this algorithm.
The algorithm that is used to generate the accompaniment is largely the same as the one used for Experiment 4 with a couple of changes. First, after completing Experiment 4 I realized I had to find a way to set the sixteenth note counter to zero every time the meter is changed. Otherwise, occasionally when changing from one meter to another there may be one or two extra beats. However, reseting the counter to zero ruins the subroutine that sends automation values. Thus, I had to use two counters, one that keeps counting without being reset (on the right) and one that gets reset every time the meter changes (on the left).
Initially I made a mistake by choosing the new meter at the beginning of a phrase. This caused a problem called a stack overflow. At the beginning of a phrase you’d chose a new meter, which would cause the phrase to reset, which would a new meter to be chosen, which then repeats in an endless loop. Thus, I had chose a new meter at the end of a phrase.
Inside pd choosemeter, we see the phrase counter being sent to two spigots. This spigots are turned on or off from depending on the value of currentmeter. The value 0 under r channel sets the meter for the first phrase to 4/4. Using the nomenclature of r channel is a bit confusing, as s channel is included in loadbang, and was simply included to initialize the MIDI channel numbers for the algorithm. In a future iteration of this program, I may retitle these values as s initializevalues and r initializevalues to make more sense.
Underneath the two spigots, we see sel 64 and sel 48, This finds the end of the phrase when we are in 4/4 (64) or 3/4 (48). Originally I had used the values 63 and 47 as these would be the numbers of the last sixteenth note of each phrase. However, when I did that I found that the algorithm skipped the last sixteenth note of the phrase. Thus, by using 64 and 48 I actually am choosing the meter at the start of a phrase, but now reseting the counter to zero no longer triggers a recursive loop. Regardless, whenever either sel statement is triggered the next meter is chosen randomly, with zero corresponding to 4/4 and 1 corresponding to 3/4. This value is then sent to other parts of the algorithm using send currentmeter, and the phrase is reset by sending a bang to send phrasereset.
As previously noted, I figured out how to connect the EYESY to wifi, allowing me to transfer files two and from the unit. Amongst other things, this allows me to download new EYESY programs from patchstorage.com and add them to my EYESY. I downloaded a patch called Image Shadow Grid, which was developed by dudethedev. This algorithm randomly grabs images stored inside a directory, adds them to the screen, and manipulates them in the process.
I was able to customize this program without changing any of the code simply by changing out the images in the directory. I added images related to a live performance of a piece from my forthcoming album (as Darth Presley). However, up to this point I’d been using the EYESY to respond mainly to audio volume. Some algorithms, such as Image Shadow Grid, use triggers to incite changes in the video output. The EYESY has several trigger modes. I chose to use the MIDI note trigger mode. However, in order to trigger the unit I now have to send MIDI notes to the EYESY.
This constitute the other significant change to the algorithm that generates the accompaniment. I added a subroutine that sends notes to the EYESY. Since the pitch of the notes don’t matter, I simply send the number for middle C (60) to the unit. Otherwise the subroutine which determines whether a note should be sent to the EYESY functions just like those used to generate any rhythm, that is it selects between one of three rhythmic patterns for each of the two meters, each pattern of which is stored as an array.
As with last month, the significant weakness in this month’s experiment is my lack of practice on the WARBL. Likewise, it would have been useful if I had the opportunity to add a third meter to the algorithm that generates the accompaniment. The Organelle program 2opFM is not quite as expressive with the WARBL as it had been in experiment 2 and 4. Changes in the FM Mod Index aren’t as smooth as I had hoped they’d be. Perhaps if I were to expand the patch further, I’d want to add a third operator, and separate the FM Mod Index into two parts, one where you set the maximum level of the index, and another where you set the current level, so you can set it where the maximum breath pressure on the WARBL only yields a subtle amount of modulation.
In terms of the EYESY, I doubt I will begin writing programs for it next month, I may experiment with already existing algorithms that allow the EYESY to use a webcam. However, hopefully by October I can be hacking some Python code for the EYESY. My current plan is to experiment with some additive synthesis next month, so stay tuned.
While this month’s experiment may not seem musically much more advanced than Experiment 2, this month has actually been a significant step forward for me. I finished reading Organelle: How to Program Patches in Pure Data by Maurizio Di Berardino. More importantly, I finally got the WiFi working on my Organelle which allows me to transfer patches back and fourth between my laptop and the Organelle. I used that feature to transfer a patch called NESWave Synth by a user called blavatsky. This patch uses waveforms from the Nintendo Entertainment System, the Commodore 64, and others as a basis for a synthesizer. However, the synthesizer allows one to mix in some granular synthesis, add delay, add a a phasor, and other fancy features.
I made one minor tweak to NESWave Synth. In experiment 2 I used my WARBL wind controller to control the filter of Analog Style, I wanted to do the same with NESWave. On Analog Style, the resonance of the filter is on knob 3 and the cutoff frequency is on knob 4. On NESWave Synth, these to settings are reversed. So, I edited NESWave Synth so resonance is on knob 3 and cutoff frequency is on knob 4. I titled this new version of the patch NES EWI. This patch me to go from controlling Analog Style to NES EWI without changing the settings on my WARBL.
As NESWave Synth / NES EWI has a lot of other features / settings. During this experiment, I setup all the parameters of the synth the way I wanted, and didn’t make any changes in the patch as I performed, although again the breath pressure of the WARBL was controlling the filter cutoff frequency. Another user noted that NESWave Synth is complicated enough to warrant patch storage, although to the best of my knowledge no one has implemented such a feature yet.
The tweak I made to NESWave Synth is insignificant enough to not warrant coverage here. Accordingly I’ll go over changes I made to the PureData algorithm that generated the accompaniment for Experiment 2. Experiment 2 uses the meter 4/4 exclusively. I’ve been wanting to build an algorithm that randomly selects a musical meter at the beginning of each phrase. While the basic mechanics of this is easy, in order to add a second meter, I have to double the number of arrays that define the musical patterns.
In Experiment 4 I add the possibility of 3/4. Choosing between the two meters is simple. Inside the subroutine pd instrumentchoice I added a simple bit of code that randomly chooses between 0 and 1, and then sends that value out as currentmeter.
However, making this change causes three problems. The first is that the length of each measure now has change from 16 sixteenth notes to 12 sixteenth notes. That problem is solved in the main routine by adding an object that receives the value of currentmeter and uses that to select between an option that passes the number 16 or the number 12 to the right inlet of a mod operation on the global sixteenth note counter. This value will overwrite the initial value of 16 in the object % 16. As I write this, I realize I need to also reset the counter to 0 whenever I send a new meter so every phrase starts on beat 1. I can make that change in the next iteration of the algorithm.
The next problem is that I had to change the length of each phrase from 64 (4 x 16) for 4/4 to 48 (4 x 12) for 3/4. This is solved exactly the same way by passing the value of currentmeter to an object that selects either 64 or 48, and passes that value to the right inlet of a mod operation, overwriting the initial value of 64. Note that I also pass the value of currentmeter to a horizontal radio button so I can see what the current meter is. I can’t say that I actually used this in performance, but as I practice with this algorithm I should be able to get better at changing on the fly between a 4/4 feel and a 3/4 feel. Also, this should be much easier for me when I play an instrument I am more comfortable with than the EWI. I have also added a visual metronome using sel 0 4 8 12 each of which passes to a separate bang object. Doing this will cause each bang to flash on each successive beat. In future instances of this algorithm I may choose to just have it show where beat one is, as counting beats will become more complicated as I add asymetrical meters.
The final problem is that every subroutine that generates notes (pdmakekick, pd makesnare, pd makeclap, pd makehh, pd makecymbal pd maketoms pd makecowbell pd makepizz, pd makekeys, pd makefm) needs to be able to switch between using patterns for 4/4 and patterns for 3/4. While I made these changes to all 10 subroutines, it is the same process for each, so I’ll only show one version. Let’s investigate the pd makekick subroutine. The object inlet receives the counter for the current sixteenth note, modded to 16 or 12. This value is then passed to the left inlet of two different spigot objects. In PureData spigots pass the value of the left inlet to the outlet if the value at the right inlet is greater than zero. Thus, we can take the value of currentmeter to select which spigot gets turned on, and conversely, which one gets turned off, using the 0 and 1 message boxes.
Now we know which meter is active, we will then pass to one of two subroutines which picks the current pattern. One of these subroutines pd makekick_four is for 4/4. The other, pd makekick_three is for 3/4. Both have essentially the same structure, so let’s look inside pd makekick_four. This subroutine uses the same structure as pd makekick. Again, the inlet receives the current value of the sixteenth note counter. Again, we use spigots to route this value, however, this time we use three spigotsas there are three different patterns that are possible. This routing is accomplished using the current value of pattern. The 0 position of this array stores the value for kick drum patterns. Technically speaking, there are four different patterns, 0, 1, 2, 3, with the value 0 meaning that there is no active pattern, resulting in no kick drum pattern. Again, a sel statement that passes to a series of 0 and 1 message boxes turn on and off the three spigots. The object tabread objects below read from three different patterns kick1_four, kick2_four, kick3_four. The value at this point is passed back out to the pd makekick subroutine. Since the possible values, 0, 1, 2, or 3, are the same whether the pattern is in 4/4 or 3/4, these values are then passed to the rest of the subroutine, which either makes an accented note, an unaccented, note, or calculates a 50% chance of an unaccented occuring. A fourth possibility, no note, happens when the value is 0. By not including 0 in the object sel 1 2 3 we insure no note will happen in that case.
While I still haven’t looked into programming the EYESY, I did revise the PureData algorithm that controls the EYESY. In Experiment 2, all of the changes to the EYESY occur very gradually, resulting in slowly evolving imagery. In Experiment 4, I wanted to add some abrupt changes that occur on the beat. Most EYESY algorithms use knob 5 to control the background color. I figure that would be most apparent change possible, so in the subroutine pd videochoice I added code that would randomly choose four different colors (one for each potential beat), and to store them in an array called videobeats. Notice that each color is defined as a value between 0 and 126 (I realized while writing this, I could have used random 128) as we use MIDI to control the EYESY, and MIDI parameters have 128 different values.
Now we have revise pd videoautomation to allow for this parameter control. The first four knobs use the same process that we used in Experiment 2. For the fifth knob, the 4 position in the array videostatus, we first check to see whether changes should happen for that parameter by passing the output of tabread videostatus to a spigot. When tabread videostatus returns a one, the spigot will turn on, otherwise, it will shut down. When the spigot is open, the current value of sixteenth note counter is passed to an object that mods the by 4. When this value is 0, we are on a beat. We then have a counter that keeps track of what the current beat number is. We then however, have to mod that either 4 for 4/4 or 3 for 3/4. The is accomplished using expr (($f1) % ($f2)). Here we pass the current meter to the right inlet, corresponding to $f2. We do thus using the value of currentmeter to select between a 4 and 3 message box. We can then get the color corresponding to the beat by returning the value of videobeats, and send that out to the EYESY using ctrlout 25 (25 being the controller number for the fifth knob of the EYESY).
In terms of the video, it is clear that I really need to practice my EVI fingerings. I am considering forcing the issue by booking a gig on the instrument to force me to practice more. I find that the filter control on Analog Style felt more expressive to me than the control on NES EWI. Though perhaps I need to recalibrate the breath control on the WARBL. I hope to do a more substantial hack next month, perhaps creating a two operator FM instrument. I also hope to connect the EYESY to WiFi so I can add more programs to it.
June has been busy for me. I’m behind where I wanted to be at this point in my research. This is due in part to not having a USB WiFi adapter to transfer files onto the Organelle and the EYESY. Accordingly, this month’s post will be light on programming, focusing on the Pure Data object route. I’ll be ordering one later this week, and will hopefully be able to take a significant step forward in July.
Accordingly, my third experiment in my project funded by Digital Innovation Lab at Stonehill College investigates the use of the preset patch Deterior (designed by Critter and Guitari). Deterior is an audio looper combined with audio effects in a feedback loop. In this experiment I run audio from a lap steel guitar.
The auxiliary key is used in conjunction with the black keys of the keyboard to select various functions. The lowest C# toggles loop recording on or off , the next D# stores the recording, the F# reverts to the most recent recording (removing any effects), the G# empties the loop, while A# toggles the input monitor. The upper five black keys of the instrument toggle on or off the five different effects that are available: noise (C#), clip (D#), silence (F#), pitch shift (G#), and filter (A#). The parameters of these effects cannot be changed from the Organelle, they can only be turned on or off. The routing of these auxiliary keys is accomplished through the subroutine shortcut.pd, using the statement route 61 63 66 68 70 73 75 78 80 82. Note that these numbers correspond to the MIDI note numbers for the black keys from C#4 to A#5.
A similar technique is used with the white keys of the instrument. These keys are used as storage slots for recorded loops, which can be recalled at any time once they are stored. This is initiated in pd slots, using route 60 62 64 65 67 69 71 72 74 76 77 79 81 83. These numbers correspond to the white keys from C4 to B5.
In this experiment I start using only filtering. Eventually I added silence, clipping, noise, and finally pitch shifting. Specifically, you can hear noise entering at 5:09. After adding all the effects in I stopped playing new audio, and instead focused on performing the knobs of the Organelle. Near the end of the experiment I start looping and layering a chord progression: Am, Dm, C#m, to Bm. When I introduce this progression I add one chord from this progression at a time.
In terms of the EYESY, I ran the recording through it so I would have my hands free to perform the settings. Again, I used a preset as I did not have a wireless USB connection to transfer files between my laptop and the EYESY. Hopefully next month I should be able to do deeper work with the EYESY.
Like Experiment 1, I had a lot of 60 cycle hum in the recording, which I tried to minimize by turning down the volume on the lap steel when I’m not recording sound. That process created it’s own distracting aspect of having the 60 cycle hum fade in and out in a repeating loop.
May has been a busy month for me. Thus, my second experiment in my project funded by Digital Innovation Lab at Stonehill College investigates the use of the preset patch Analog Style (designed by Critter and Guitari). To be specific, I am using a WARBL wind controller with EVI fingering to control the patch. I am using the breath control on the WARBL to control the cutoff frequency of Analog Style (using MIDI controller 24).
Due to the busyness of the end of the semester, this experiment features no original programming on the organelle. However, I did create a program in Pure Data to create accompaniment and drive the EYESY. To accompany the experiment, I used the H.E.A.P, the Housatonic Electronic Algorithmic Philharmonic. This a fun, frivolous name I’ve given to a small, battery powered synthesizer / sampler setup I’ve assembled for live performance. It consists of three synthesizers / samplers: a Volca Sample 2 (which provides 10 channels of late 1980s style lo fidelity digital sampling), a Volca Keys (which can be used as an analog monophonic or 3 note polyphonic synthesizer), and a Volca FM 2 (which is a clone of the 1980s classic, the Yamaha DX7, the best selling synthesizer of all time). I’ve also begun to think of the EYESY, as we’ll see later, as part of the H.E.A.P.
I won’t go into great detail about the program that generates the accompaniment for Experiment 2, as I have other blog posts that go into detail about various algorithms included in the program. Ultimately, the program is intended to generate relatively generic, but fairly usable R&B esque slow jams. The music is in common time using sixteenth notes. The portion of the program including and beneath % 16 (mod 16) ensures that the resulting music will have 16 pulses per measure. Likewise, the instrumentation and musical patterns change every four measures. This is enabled by the part of the program including and beneath % 64 (four measures of sixteenth notes adds up to 64).
The Volca Sample is being used to provide the drum beat, and some string pizzicato (see pd makepizz). The Volca Keys provides some synthesized bass patterns that run two measure loops. The Volca FM provides four chord, four note chord progressions that repeat every two measures. To create these chord progressions I used some music programming techniques that I’ve covered in a previous blog entry, though in this experiment I am use the brass friendly key of G minor.
One of the newer programming tricks I used in this program is an algorithm designed to drive the EYESY. While the EYESY generates hypnotic, interactive video animations, left to its own devices it can get a bit repetitive fairly quickly. In order to generate anything that seems even remotely dynamic some one needs to perform the EYESY by rotating its five knobs. This is an impossibility for any performing musician, save for a vocalist. Thankfully, we can do the equivalent of turning the knobs on the EYESY through MIDI using controllers 21 through 25.
The algorithm I’ve created to drive the EYESY is designed to make slow, evolving changes. To control these changes I’ve created a table called videostatus. It consists of five positions that contain a one or a zero to denote whether changes should be make or not made to a given knob during the current four measure phrase.
The subroutine pd videochoice is triggered at the beginning of every four measure phrase. It generates five different random numbers that are either a zero or a one. These results are then stored in the table videostatus.
The subroutine pd videoautomation updates the knob positions on the EYESY once every sixteenth note. It is passed the current sixteenth note number modded by the number 224, which corresponds to 14 measures of sixteenth notes. The subroutine contains five nearly identical columns, each of which corresponds to each of the five knobs on the EYESY. First the algorithm checks the current states of each of the five positions of videostatus table. When that value is one, that allows the current sixteenth note number to pass through the spigot. This value is passed through an expr statement that displaces the sixteenth note number. The column corresponding to knob one is not displaced, but each subsequent column is displaced by one measure (16 sixteenths), which is then modded to stay between 0 and 224. The statement moses 112 is used to determine whether we should be counting up to 112, or counting down to 0. This is accomplished by having numbers that are greater than 112 to pass through expr (224-$f1), which causes the result to get small as the input value is increased. The result of this is then passed to one of the five controller values (21-25) on MIDI channel 16 (the channel I’ve set my EYESY to).
Since I went over the mother patch in the previous experiment, I’ll start with going over main.pd for the Analog Style patch. We can see that knob one controls the tuning of the patch, while knob two creates an offset frequency for a second oscillator. The third knob sets the resonance of the low pass filter, while the fourth knob (the one I am controlling using the WARBL) sets the cutoff frequency of filter. To learn more about what low pass filters are, check out my blog entry on Low Pass Filters in LogicPro. However, to summarize briefly in relationship to the WARBL, when the amount of breath coming through is low, that in turn sets the cutoff frequency to be low as well, resulting in less sound (and only low frequency sound) to come out of the Organelle.
We can also see that this patch allows for sequencing when the aux button is down. However we will not go through how sequencing works today. We will however go into simple, which is the subroutine that creates the sound. We can see two oscillators, blsaw, in this subroutine that generate sawtooth waves. For more information on subtractive synthesis waveforms (including sawtooth waves), check out my blog entry on Subtractive Synthesis Waveforms in Logic Pro. One of those two blsaw oscillators is modified by the offset of knob two. The mixture of these two oscillators is passed to a low pass filter, moog~. This object also receives a center frequency to its center inlet, and a resonance value to its right most inlet. The outlet of this object is then attenuated slightly, *~ .75, before being sent to the subroutine’s outlet.
Again, I’ve found that the accompaniment generated by experiment.pd to be generic, but also fairly usable. It should be relatively easy to change the tempo, phrase length, or any number of musical patterns to create music that is stylistically different. Also, I enjoy slow evolving nature of EYESY generated video. I feel that turning on and off changes to various combinations of the five knobs add a degree of subtlety that aid in the dynamic nature of the video.
I am disappointed in my performance on the WARBL. I am still getting used to the EVI fingering on the instrument, so there are some very sour notes from time to time. However, I am very pleased with the range of the WARBL, as well as the subtle breath control the instrument provides. The fingering makes jumping octaves and fifths very easy. In future experiments I hope to get into hacking existing Organelle patches. I also plan to come up with variants of the videoautomation algorithm to create more sudden, less subtle changes to the EYESY’s settings.
