Mic Slide: A brass slide with an AKG 411 condenser microphone attached. Produces a very direct sound when rubbed against the strings or held on a string while hitting the corresponding key. 

NOTES: 

 64 They had bought a Helpinstill system at a venue in Oslo and I also heard that Christian Wallumrød used it for prepared piano amplification, so I decided to check it out. The Yamahiko was found by trawling the internet. The internet and the jungle telegraph are my main methods to discover new equipment. [back]

 65 http://www.inagrm.com/accueil/concerts/lacousmonium [back]

 66 http://music.ece.drexel.edu/research/mrp [back]

    https://www.youtube.com/watch?v=f79d_oVqv4Y

 67 www.testingtesting.org/synaesthesia/EMO/index.html [back]

 68 In English: Murder & Design. A design company/collective specializing in interaction design, with Sven Håkon Voldum, Anders Ekroll, Simon Søgnen Tveit and Per-Johan Sandlund. [back]

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This section consists of three effect chains, based on 18 different pedals. The input is one mono signal from a Yamahiko Piano Pickup or a Helpinstill pickup system. The Yamahiko is placed under the grand piano, and attached firmly to the resonant bottom using pressure from the wooden beams holding the grand piano construction. The Helpinstill is mounted inside the case, with no use of screws or adhesives. This is a gentle way of using contact microphones/humbuckers on a grand piano, because it is not involving any mechanical parts, tape or glue that may harm the lacquer on the wood.

Chain A – Multi FX:

A:1 Earthquaker Devices ToneJob EQ: 3 band EQ with 20 db. input boost, useful when working with unpredictable input signals regarding spectrum and dynamics. Each band can boost or cut the actual signal fivefold the input level, which makes this EQ powerful for signal shaping. 

I use it before the other effects because I want to control the piano pickup sound before I send it to the other devices. The sound can be very different from instrument to instrument, and is also depending much upon microphone placement. This calls for a frequency control option to ensure a good starting point for the processing. This device alters level and spectrum of the sound, and relates to the trail brightness – darkness. 

Other tests: I have tried to route the Yamahiko through my digital section (Ableton Live) before the pedal section. Then I could do a precise digital graphic EQ of the signal before my effects. I turned down this strategy because of latency and more sources of error coming with another link in the signal chain. I often found myself troubleshooting my complex pedal section, while the error was in the patching or routing of the digital section. This was more complex than necessary, so I decided to separate these systems to make the setup more transparent. I still patch the pedal section back into the digital domain, but now I do it via a DI-box split just before the amplifier. Then I will know if my pedals are working before I involve the digital section. This eases the process of troubleshooting a great deal. 

A:2 TC Bodyrez: Pedal designed to warm up piezo pickups on acoustic guitar. It also has a circuit that controls feedback to some extent. 

I use it because I can get a little more level in the mid and low area before feedback. 

Other tests: To my knowledge, there are no other devices with this combination of filtering, subtle compression and feedback-suppression. I have tried using compression devices like the Analogman Comprossor and the Xotic XP Compressor. The problem with compression in an instrument like mine is that it will lower the level when I am playing and raise the level while I am not playing, increasing the risk of unwanted feedback when I am not making any sound. Because of this I decided to exclude pure compression processes in my pedal section.

Using a designated feedback-suppressor in combination with a compressor might work even better for the same purpose, but the options available were too big for my setup. 

A:3 OTO Machines Biscuit: An advanced bit crusher (biscuiting) with some effects and a filter section. It allows for muting and inverting each of the bits in an 8 bit downscaling process. It can add wave shaping, delay, pitch shifting and modulated filter processing to the signal. OTO also has a synth generator and the possibilities of storing presets. 

I use it because of its many options on bit crushing the piano signal. I like this effect, but I often find it too bright or too prickling. This device allows for filtering and many nuances on this type of processing, and it also has the capability of creating distortions, tremolo and noise effects with a digital vibe. These sounds have a brightness that make them cut through in complex soundscapes, a feature I find useful. The possibility of saving presets is good when switching rapidly between different sceneries. This versatile device can alter spectrum, granularity, gait, pulse, structure, onset, tail and dynamics of the original sound. 

Other tests: I tried to work with bit crushing in the digital section, but this is a process I find quite sensitive to latency, so I moved it to the no-latency pedal section. I also liked the Bugbrand Bugcrusher and the WMD Geiger Counter, but I found the OTO a bit more versatile for my setup. 

A:4 Earthquaker Devices Rainbow Machine: Can pitch shift the signal into two different pitches, and alter the lag between the original signal and the pitched signal. It also has a ‘magic’ function, used to make various sound cascades of dystonic material. 

I use it because I like the effect when I detune the piano slightly, putting the pitch adjustment knob just a few millimeters off centre. This is my specific use for this box, but used fully it may abstract the sound completely, alter gait and structure of the sound, besides affecting the pitch and thereby also the spectrum of the sound.

Other tests: The Strymon Blue Sky does something similar, but then as a part of the reverbed signal (wet-signal), making the pitched representation more diffuse. A similar effect can be obtained using a modulated short delay mixed with the dry-signal, but I find this strategy a little less direct. There are lots of pitch shifters on the market doing quite similar alterations to this parameter, yet they react slightly different to setups and inputs and therefore they have to be tried in the specific setting to judge their effectiveness and response. 

A:5 Moog Ringmodulator: Altering the timbre and pitch of the original sound by modulating that sound with a sine wave, multiplying the waveforms. 

I use this device because it is the warmest and least harsh ring modulator I have found for the piano sound. This device can dissolve pitch completely or be used very subtle. It also alters granularity and spectrum of the sound. 

When ring modulating, I find that the analogue devices produce a sound closer to my poetics than the digital ones. I think that the digital devices are too correct and too cold. From my point of view, this process benefits from an analogue circuitry. 

Other tests: This process is also tested in the digital domain, but I didn´t find software that responded satisfactory. With ring modulation and other distortion processes especially I search for devices that respond distinctly and with small changes in details based on my action on the keys. I especially look for the response in the onset phase. Something more than just a predictable response to an input sound. When the sound result from a process feels like an extension of the arm I am onto something. CoPilot Android and Zvex Ringmod are other interesting hardware devices for this use. 

A:6 Red Panda Particle: A versatile pedal, using granular synthesis processing to chop up the original sound and process the result with various delay or pitch processing styles. The grain size and wet/dry mix are adjustable. 

I like this pedal because it is capable of altering sound surfaces, but also the musical material due to its randomness and unpredictability. It can depict a cloud of shattered piano sounds or a drunk copycat. I like the directness of the stomp box interface, which is difficult to obtain in a computer setup. This device works with gait, granularity, pitch, structure, pulse and tail of the sound. 

Other tests: Software tools based on granular sound processing like the Granulator by Robert Henke, M4l Granular-to-go, M4L Loop Shifter and the IRCAM IM Mover. All these have been a part of my setup. I have included a random granular playback feature in my own programming of the 4trackLooper in Max. The Hadron device in my setup also works with this technique. All these granular devices have different architecture and by that a very different sound and functionality. This makes it difficult to judge their quality, as they are different tools for different uses. I would recommend testing a broad variety of devices in a category like this, searching for what music the different units can produce from a particular input sound. 

A:7 Bugbrand PT delay: A delay with low pass filter and drive on the delay signal, and hi pass filter/boost on the input signal. Can receive CV-in, altering the speed of the delay using LFOs or expression pedals. 

I like this delay because it has a dusty, lo-fi delay output that may be filtered, and I like the opportunity to take out the dry signal completely from the mix. The LFO controlled delay speed can create big pitch alterations with different modulation shapes.

Other tests: I have used the MoogerFooger Analog Delay with a Moog VX-351 CV Expander. This setup is doing much of the same as the one I currently use, but takes up a little more space in the pedal board. In the start of the project I used the Boss-RE20 Tape Echo emulation pedal, which I found a little too simple regarding modulation possibilities. Also the Roland 3000 SDD pedal has been in use in my setup, this delay is digital with more modulation options and sounds slicker. I often use this device in the studio, but I find it a bit too heavy to take on the road. 

A:8 Earthquaker devices Arpanoid: This device arpeggiates the sound sent into it with adjustable scales, range and rate.

I like it because this is a strange effect in my music, usually related to techno and beat-based genres. In my instrument this device creates a strange echo with its pulse based processing of the original sound. It adds an artificial/strange/twin peaksy factor to the sound. Alters pitch, pulse, structure, tail and spectrum. 

This is the only device I have tested in this category, and I fell for it immediately. I like the simplicity of the pedal. There are pedals like the AdrenaLinn and other sequencer based effects that would be interesting to try out in the future. 

A:9 ZVEX Tremorama: A tremolo that varies the level on the sound based on eight adjustable steps with a speed control, in which the playback-directions may be randomized and varied. Makes vacillating dynamic changes on a sound. That is what I particularly like with it.

Other tests: This is the only analogue stomp-box to my knowledge doing randomized tremolo. The M4LBufferShuffler that I use in Ableton is also doing randomized alterations to the dynamics. Other sequencer-based effects might also be useful for these kinds of processes. 

A:10 ZVEX Lo-Fi loop junky: A 20 second looper cutting off frequencies above 2.6 kHz, with compression and tone-adjustment.

I like it because it sounds dusty and old. The sound reproduction is drifting in pitch and the surface of the recorded sounds are rough in a humane way. Alters structure and surface and is one of the memories of my instrument. 

There are many loopers available. This looper is not chosen mainly because of its looping functionality, but for the original sound quality of the looped playback. 

Other tests: Nothing like it

A:11 Strymon Blue Sky Reverb: A digital reverb that emulates spring, plate or hall reverb. May have an extremely long decay, almost like freezing the sound, and have the opportunity of modulations to the reverb pitch. Also has a feature that generates shimmering reverbs.

I like this reverb because it is not too bright, and it has a good frequency response towards the piano sound. Many reverbs enhance too much low-mid, making the sound output muddy and unclear. In addition, I like that the parameters of decay and modulation can be pushed towards the extreme. Modulates spectrum, grains and tail of the original sound. 

Other tests: I have two other reverb processes in the digital section with the Valhalla Vintage reverb, and the Altiverb EMT Plate convolution reverb. With these three options I have covered a range of reverbs ranging from the warm and short to the long and artificial. In addition to this, I have worked with the Ekdahl Moisturizer and the Vermona Retroverb, both mechanical spring reverbs. I like both of them, and especially the Ekdahl where the spring could be played. Yet these two devices had problems with travelling, as the springs are easily destroyed and they tend to pick up fast on local power issues. I had to cut them from my performing setup, but I still use them for studio purposes. 

A:12 Source Audio Programmable EQ: 7 band graphic EQ to adjust the frequency spectrum precisely.

I use this in the end of the chain to adjust problematic frequency areas after processing. Like the pianos, every amplifier is a little different and also the rooms may contribute to a bigger accumulation in certain frequency areas. This device is a transparent and clean EQ to remove energy from certain bands, and it has the opportunity to store settings. 

Other strategies: There are other stomp boxes with multi-band parametric EQ that can work for this purpose, but this is the only graphic EQ in stomp-box version I have found. 

Chain B – Extreme Distortion

B:1 Boo Instruments Boost: A regular clean booster needed to produce enough level for the next pedal.

Many boosters would work for the same purpose. 

B:2 Squarewave Parade Pollen: A signal noise generator and distortion with audio in. When audio in is being used, the internal noise and pitch generator is muted, and the distortion processing circuit is used for the input sound.

This box can create extreme distortion and it has a gate function when there is no audio in. The result is that I can play extremely loud without any feedback, which is unusual when distorting acoustic sound. Artificial and still grainy and human. I use it with a volume pedal to be able to control the onsets of the sound.

Other tests: None like this. It sounds like a blend between a fuzz and a wave-shaper, and it is in my opinion a unique device due to its headroom before feedback. 

Chain C – Stutter 

C:1 HEXE Revolver: A device that in different ways creates stuttering effects to the sound input. May be controlled in real time but is also working on its own in auto mode. 

I like this device because it generates unpredictable responses to what I play. I also like stuttering and uncertainty in music, the effect is creating a friction, questioning what is being said. 

Other tests: It is in the same family as the Teaspoon, but much more unpredictable in terms of sound output. I haven’t tested anything else like it. 

C:2 Squarewave Parade Teaspoon cream and sugar: An instant looper or window sampler, able to hold small parts of audio before a hold-button is activated. The hold function can be locked. When it is pushed, the pedal plays back the last bit of audio, depending on the adjustable sample size. The speed can also be adjusted. It works like a real-time stutter pedal, and the sampling function may be activated by a dedicated foot-pedal. A very rare pedal on the market. 

I like that it can freeze a moment, take out a small incident in the music and loop it. Because the looping time is very short (around one second) it is not working like a normal looper but more like a freeze-effect. It is useful to create stuttering decays on a tone, or to comment melodic phrases or chords. It has a distinct lo-fi sound quality.

A combination with the HEXE Revolver can create an illogical and random output that I find useful, feeding my music with random inputs in real time. 

microphones, processing and amplification

When I started the process of attaching the eBows to the grand piano, I discovered another fantastic project by Andrew McPherson, called the Magnetic Resonator Piano. He has attached electromagnets to every string on a grand piano. These can be controlled from the grand piano keyboard, and adjusts automatically to the tuning on the piano ⁶⁶. Unfortunately, this system is not for sale…

My device is made using a Marxophone, an Arduino UNO card, MosFet micro controllers and modified eBow electro magnets. A Max patch is used to control the voltage on the eBows. In this version, the instrument is not sturdy enough, and broke down as a result of travelling. This instrument will be developed further in upcoming projects. 

modules and patching

People often ask me how much of the equipment I build myself and how much of the computer programming I do. 

Every module is patched with the other modules in the system, either directly via audio cables, control signal cables (Midi and USB) or indirectly via recording devices, loudspeakers, audio bled into microphones or on a more metaphysical level that relates to how I am working with multiple devices simultaneously in the music making process (patched up in my head). Where the module is placed physically and the haptic layout is also important. The connection of equipment regarding audio signals and control signals, along with the physical placement of the modules in the instrument and the way the modules are played and layered in the music using different outputs; this is essential aspects making the HyPersonal Piano my instrument. 

I am not an engineer in electronics nor music technology, so all of the hardware I have to either obtain ready made, programme myself or collaborate with someone that customizes solutions.

A large part of my work is to stay updated on what gear is made in small garages or big music tech assembly lines. The Lightning Mountain and the Marxuino are made from scratch with collaborators. I have learned to programme in Max MSP, and I have done some patches alone and some in collaboration with composer Christian Blom. The programming for the Lightning Mountain is done in C+ by the Drap&Design team. I have also used some programming by my second supervisor Øyvind Brandtsegg. These are done in C+ and adapted to VST-format using Csound and Cabbage. 

I regard any device or software as a module of the system which constitutes my instrument. All devices and software have a sound of their own and can be altered to a greater or smaller extent. When I test a module for a possible integration in the instrument, I look for what it does sonically, the possibilities of alteration and how this device interacts with the rest of the instrument. This is very important to me. It is often in the complexity of the way these modules work together sonically that the new sounds occur. 

Modules with certain attributes can be connected in several ways, and tweaked to get the desired outputs. I see it as a valuable part of working with equipment that others have made to adapt these to my sound and playing, as well as making devices from scratch. Sometimes, with open source software, one can also adapt devices that others have made to own needs. The combination of these strategies and the poetic reflections involved in combining the different modules is making sure that there will be a HyPer(sonal) Piano music. 

Contact microphones left and right: 2 slightly different piezo pickups plugged into Ableton and panned left/right. Used to play on the strings or the metal board to make rhythmical patterns, often through the stutter plug-in. Sometimes held against the body of the instrument while playing, or used with external sound sources like the Marxuino (See ”marxuino” on page 149) or the Bugbrand Boardweevil. 

Radial DI boxes: For the computer output I use the Radial JDI stereo DI. It is a big gain to have a designated DI box for a complex setup like mine, and this box is dealing with high inputs and ground loops humming.

For the pickup distribution, I am using the Radial JDV DI. It has steep-less ohmi-adjustment on the input, tailor made for piezo pickups. It also has several options on signal splitting, a lowpass filter and input selection. This box allows me to switch between the Yamahiko and the Helpinstill system. 

Bringing my own DI boxes is one of the most important factors to ensure stability and an instrument sounding as good as possible. 

Palmer PLI-01 line isolation box: Small device used to remove occasional hum from ground loops in the pedal section. Local circumstances on the venues regarding current, grounding and power distribution often create problems. This box deals with that in most cases. 

Other tests: For a period I used the Radial ToneBone for pickups: It is a heavier box that can blend two piezos and also do filtering of the signal. I found that for my use, the blend function was not necessary, so I started using the JDV instead. This being said, the ToneBone is a solid problem solver when working with piezo microphones.

(from jöK & seasicK #6, 16 September 2013, NMH)

Morten Qvenild: The HyPer(sonal) piano project

Currently in section: Technical manual

These models and the following technical descriptions describe the instrument as of June 2016. The instrument will continue to evolve after this, and has evolved up to this point. Other devices that have been tested but not included in this setup will also be described in the following texts. 

digital section I, Ableton Live

I don´t aim to describe exact technical specs of each device or patch. I rather try to describe what the devices do musically and how they relate to the trails and poetics of my project. 

Workspace and workflow

I started using Ableton Live when I initiated this project. The choice was based on the need to import software tools from the studio setting into the performing situation. The option to create my own devices in Max and utilize them in Max for Live (M4L) also influenced my choice. Ableton has a series of stable audio and midi tools that work on a low CPU load, bread-and-butter processes like compression, filtering, delay and transposing. The possibility of having a digital interface that could incorporate third-party plug-ins that I had used in the studio, my own programming for live purposes and the stable Ableton midi and audio architecture, made Ableton seem ideal for the project. 

This being said, the introduction of custom made patches through Max4Live, alongside other external programming sources, destabilized the programme a lot. I have been in touch with the Ableton support team with problems frequently, most often related to third party plug-ins. It took about 16 months of work on restructuring and troubleshooting the Ableton setup before I had a solution stable enough to go on stage. The problem with instability is not related to the Ableton architecture, but mostly to faulty plug-ins causing memory leaks or conflicts between different software tasks, complicated by third-party software. 

Software choices are important for the workflow. The workflow is essential to how the music turns out. A workflow with too many obstacles in terms of instability, needs for file and software conversion and frequent rounds of troubleshooting and restarting take focus away from the musical process. To accomplish a good workflow, I need a complexity in the setup that can match and challenge the complexity of my music making. The setup needs to be stable and sturdy, not letting technical problems or a bungled workflow get in the way. 

I took an extensive course in Max as a part of my project. This made it important for me to be able to integrate my work on that platform into the DAW (digital audio window) of my project. When I use the two computer setup, I use my programming in full-scale through Max. In the setup with one computer, I am using the same programming ideas, only reduced, through M4L in Ableton. The reduction of Max programming into the M4L environment is forcing me to find the essence of the processing, deciding which functions of the patch I need and which to exclude. This clarification often leads to changes in the original patch too. 

Another important concern I had when choosing platforms and building a workflow was the midi routing possibilities. A main aspect of constructing an electronic instrument is the interface, assigning buttons, pots and sliders to different parameters in the software. New music often demands change of these setups quite frequently, and I need to be able to do these changes very swiftly to not obstruct the musical workflow. Other DAW´s designed for tracking like Reaper, Pro Tools, Cubase and Logic have a structure making midi-routing a more cumbersome process. 

Computer bullet points

• Always have the latest updates of software and all plug-ins. Check compatibility with your operating system. 
• Involve the software-manufacturers support team. There is a lot to learn from those who wrote the codes. 
• Use the DAW´s internal plug-ins when possible to save on CPU-load. 
• If crashing, try to remove suspicious tracks one by one, and see if the problem persists.
• Use buses, not inserts if possible. 
• Check the CPU-meter when adding a process to see what the process demands regarding data-power. 
• Use RAM- and CPU-logging software like the Activity Monitor to see what is going on and reveal memory leaks and other abnormalities.
• Make sure the computer is not too hot.
• Use SSD disks. 

My computer Specs

MacBook Pro, 15-inch, Mid 2012

Processor 2.6 GHz Intel Core i7

Memory 8 GB 1600 MHz DDR3

Soundcard: RME UCX, RME BABYFACE

Midi: Novation SL MK2 25 keys with TouchKeys, Midisport 2x2, Kenton Killamix Mini. 

Ableton track setup

This Ableton setup uses about 28 % CPU, and utilize about 1.3 GB RAM when running normally. With this setup, I can play on 128 samples (appr. 5 ms) latency. In my instrument, I´ve found that anything under 8 ms. latency is acceptable. 

digital section II, Max

In settings when I am not playing the Personal Piano material, I often use a second computer in my setup. This computer executes two patches in Max. The first tool is a four track recorder and playback machine with modulation and random playback and time stretching possibilities. The second tool is a midi-recorder and playback machine with randomization of playback notes, orders and durations, made in collaboration with research fellow Christian Blom. I send audio to this CPU from the first CPU using Adat optical. The midi is routed from the first CPU to the second CPU using a software called Bome´s Midi Translator.

4 track recorder/random playback/time stretch 

The device is controlled from the computer keyboard and mouse, but single parameters may be set up to be controlled from a midi-controller if needed. In this case for my setup, the keyboard/mouse setup was the most convenient and compact.

The buffers of each track can record up to 1 minute of stereo audio. The buffer can be larger or smaller by changing arguments to the buffer-object in the program. The waveform windows of the programme are dynamic, and adjust visually to the length of the recorded sound file. Each track can be deleted by pushing CLEAR once. Controller types for editing and playback of the sound file are chosen in the toolbar for each waveform window.

Regular playback, forward/reverse and stop can be chosen with individual keys for each track. The playback speed can be altered with a slider, and fine tuned with a dial connected to the slider. Panning and volume are set individually on each track. In addition, the playback speed on each track may be regulated to create a tape-recorder emulation with smaller or larger variations in tape-speed. The modulation is controlled by a sine wave, where speed and rate are adjustable. 

Each track has a random playback function, inspired by granular techniques. The starting points are chosen randomly: Length of playback and speed of the starting point selection can be tweaked. The selection speed can be regular or varied, depending on settings. 

If I create something with the patch I want to keep, it´s easy do a mix down of this material. The TimeStrech processing is activated with the recorded mix down. Timestretch can be varied in tempo, clip length and clip start. The time stretching affects the length of the recorded clip without affecting its pitch. 

The master from the device goes through a 4 track EQ. The Q-value for each band may be altered and the EQ is edited with the mouse. 

There are clicks when playing back with the random function. This is on purpose. I like the rhythmical random function of the clicking sounds. 

The keyboard controls are written in the patch window. Audio must be turned on and interface chosen by clicking the adc-object on startup. 

P- code (locked):

The remix I did of a Saskia Lankhoorn concert was done using this device a great deal. This remix is presented under the chapter on ”comfortable – uneasy” on page 60. I prepared for this remix using her record with the same compositions as she played on the concert. I recorded passes with this device, tweaked them and mixed down materials I was happy with. I created a prerecorded sound bank for use at the ‘live’ remix. 

Max Midiloooper

This device records incoming midi signals when hitting the record button, and stops using the stop-record button. All buttons and dials are routed out to a Kenton Killamix midi-controller. There is no length-limit to the recording. 

The recorded midi part can be played back as is or it can be played back with a cut-up function, choosing random starting points in the file. The length of the clips are adjustable, and the length and pitch of each tone can be gradually randomized using a slider for each parameter. 

This patch have the same functionality when used in Max or Max4Live. 

I think of the device as a remix machine. When I record midi of a longer stretch, I can stop playing and work with this device, sending the midi alterations into a sampler with a different sound. What I played on the piano is represented in a new sound. Further, I can alter the tonality and rhythmical aspects of the stretch to a degree that the resemblance to the original stretch disappear. This is creating musical material and a new input to the playing situation which I can use for further improvisation. 

P-code (open):

In this pass the midi is being used with the midilooper device, sending the processed midi into a sampler with a Casio sk-5 sample. 

pedal poetry

ringmodulator

bug brand PT delay, CV modulated

bodyrez

tremorama

mission delta III

boost

squarewave parade pollen

revolver

squarewave parade teaspoon cream and sugar

rainbow machine

arpanoid

tonejob

red panda particle

biscuit

lofi loopsampler

blue sky

broadcast

echorec

These are the names of the 18 pedals in my setup.

fx pedal section

other hardware

RME UCX Soundcard: A sound card with 8 analog in/out, and digital I/O. Good sounding converters and stability. The right sized sound card for my setup. 

I have also tested and used MOTU Ultralite Mk3 and Metric Halo ULN-2 for a longer period with good stability. I chose the RME card due to its architecture of inputs and outputs, and the fact that it can send CV out through the Phones output (unofficially).

RME BabyFace Soundcard: 2 analog in, 4 analog out, with ADAT or SPDIF I/O.

I use this second sound card when I use a 2 computer setup with my custom made recorders. 

Electroharmonix 2880 4 track looper: A hardware looper with SD-card storage for larger chunks of audio. Can store four tracks mono or two stereo. Can trim audio playback speed ungraded, pitching audio down or up one octave. Easy to control in a live situation. 

I use this hardware box mainly because it solved lots of computer problems. I used to have a similarly featured four track looper in Ableton Live, but this was a third party plug-in, and it caused major memory leaks making Ableton request up to 112 Gb of virtual RAM when I let it run overnight. They had never seen anything like it at Ableton Support.

Storing large audio chunks in computer RAM may cause problems and slow other processes, so I decided to prioritize stability before function and weight, moving this process to this hardware. All audio that I produce may be routed to this looper via Ableton. 

Other options: Digitech JamMan, Boss RC-30 or 3000, Electro-Harmonix 22500 or 45000. 

OP-1 Synthesizer: A small synth with many features. Roughly, it is divided into a recording section for recording internal or external audio, a synth section and a drum section. There are also several sequencer functions. 

I generate midi from my piano using the Moog Pianobar. These midi-signals can be routed to the OP-1, and I often use these signals to dub the piano sound with more noise-related material from this synth. I like its sound quality which is not slick. It is easy to work with dystonic materials on this device. I also generate midi from the ‘tombola-sequencer’, a feature playing back a pool of the latest input notes in random order and with irregular timings.

Bugbrand Boardweevil 2012: Two oscillator PCB design noise box/synthesizer with circuit bend copper panels and distance-sensors. 

This device is quite difficult to control, yet I like its ability to create irregular clicking pulses and sounds resembling a bird reservoir. It also has a small loudspeaker, so it can work as a little ‘acoustic’ synthesizer inside the piano case.

TC Helicon Voicelive 2: Voice processor/vocoder/harmonizer. Fully programmable with preset storage. Receives midi from Moog Pianobar. 

This device solved lots of problems with the vocal solutions when performing live. I tried to process vocal in Ableton, but it was never good enough regarding stability and headroom before feedback. 

Other hardware devices that has been tested in the setup are: DaveSmith Mopho (for bass used live with sPacemoNkey), Subtle noise maker & Weird sound generator Noisebox (solo), Eowave ribbon 2 synth (used with sPacemoNkey), BugBrand CubeWeevil noise box (used occasionally), Elektron Sid Station (used with sPacemoNkey), Moog Voyager (used for bass on The Karman Line), Folktek sound field 1 (used for studio processing on The Karman Line and Personal Piano), Folktek Insectan (used in studio) Casio SK-5 sampling keyboard (used for sample recording and playback in Ableton), Waldorf Blofeld (used in an early stage of the project). 

Mic Bundle

Taking in this view of the piano, as a surface that can be divided into a limitless amount of small spots, a different approach to recording and amplifying arises. This approach origins from working with Norwegian sound engineer Asle Karstad, and has continued evolving through working with the engineers Daniel Wold and Ingar Hunskaar. 

Every spot has a slightly different sound profile beaming out of it. Spots far away from each other may differ more than close ones. Further away from the keyboard we might find more mid- and low-end. Closer to the keys the sound is brighter, and spots near metal parts may have more body. This varies a lot from instrument to instrument, depending on the construction. One method to find your spot is to listen closely under and above the instrument, knock it lightly with a knuckle to check the resonance spectrum at different spots. Another method is to get someone to play the piano and move the microphone around while listening to it in a headset. 

Microphone philosophy 

A turning point for me in my project was when I realized that the use of microphone wasn´t a tool to amplify the sound that I heard from the grand piano directly, and then colouring it.

The search for a hi-fi piano sound caused lots of problems with unwanted and uncontrollable feedback. The solution to these problems was the method of spot-amplifying and reconstructing sound. 

The piano is a big resonance case, and all the sounds recorded by a microphone and played out in the room is being recorded again by the same microphone when the sound from the loudspeakers are coming back to the resonance case. In all amplification there are sound loops occurring, and they create difficulties. If the bounce back from the room is too large we are amplifying the room, not the direct sound. 

The attitude of reconstructing sound and not reproducing it 1:1 was a revelation, and it gives a flexibility on what processes I can do in the instrument. This strategy also makes it easier to control and use feedback for musical purposes. 

I experience that these strategies work very well for lo-fi processing, a strategy where I move the processed material away from the original sound. I work with reductions of the total sound, giving it different characteristics with the use of processing. For processing that needs access to the full-bodied piano sound, I sometimes use samples from Kontakt triggered by the Moog Pianobar. This strategy keeps the piano sound intact while avoiding feedback. 

Distance

The normal way of amplifying and recording grand piano is to place a pair of microphones about 0.5 to 1.5 meters from the instrument arch. The model for this approach is our ears: we try to recreate what we hear as a total output coming from the instrument, using two good microphones. They capture even more of the spectrum than our ears are capable of. To me, this method does not work when the sound is amplified, because the amplified sound feeds back to the microphone and creates a sonic round-dance. 

Our approach in this project is to move in closer, letting the amplified sound consist of a reconstruction where the sound from two to four smaller spots of the piano is equalized and mixed. The further we move the microphone away from the piano, the bigger area of spots we will capture, often resulting in a broader spectrum with more overtones. Yet when we move away, we capture more of the room’s characteristics and the other sounds in the room. If we work with other performers playing on higher volumes, this might be a problem. The balance between distance and sound quality must be found depending on the performance-setting and the way we are going to use the sound. 

When amplifying the ‘acoustic’ piano sound alone through a PA-system or on a recording, we might live with a higher degree of sound leakage from the surroundings and room attributes coloring the sound. That means we can work with more distance between microphones and instrument, and more open microphone characteristics (omni or 8-figure). 

Pickups

The amplification of the piano sound has been a main task, problem and issue in this project. The piano is a grand resonance case, and the attempt to amplify and process it loudly often results in grandiose feedback problems before getting a desired level. To solve this issue I had to find microphones that deliver high output without feedback. I tested a lot of varieties and ended up with the Yamahiko Piano Pickup and the Helpinstill system as the best solutions ⁶⁴. The sound from these pickups are fairly natural, and have lots of low end without creating problems. The Yamahiko, a contact microphone, has more ‘wood’ in the sound. The Helpinstill, which is a Humbucker has fewer overtones and a more ‘electric’ sound, almost like a Yamaha CP-70. The isolation in the Helpinstill system is very good, and bleeding from other instruments is never a problem. This makes it ideal for settings with high volumes. The Yamahiko is a bit more exposed to leaks from other instruments, so I use this in solo or low-volume settings. The Radial JDV DI has an option of choosing between these two pickup systems when both are mounted. 

If the microphone signal I use for processing captures the sound of other performers on stage or studio, I will end up processing the whole group through my system. This can easily become a chaotic sonic situation, adding unwanted processing to sound sources other than mine. My aim when I process a signal is to capture it as isolated as possible. 

I also have to work on manipulating the sound to reduce feedback issues. I have an EQ right after the microphone preamp to take away unwanted frequencies before processing, and I have another one before the amp-stage to adjust the frequency spectre of the processed sound. The last EQ should have more bands and be more precise than the first to be able to remove narrow problem areas. In addition I use a TC Electronics BodyRez box to enhance low mid without getting feedback. There is a built in feedback destroyer in this device which make the output slightly fatter and louder. This is for the pedal section… I split the microphone setup into the digital section and uses digital parametric EQs before the digital processing. The equalizing must often be tweaked when I move between rooms, due to different acoustics. One important aspect to remember: if the house sound engineer uses the same pickup to get the unprocessed sound out in the PA, you will work on the same problem frequencies. It is best to use separate microphone setups for the piano sound in the PA and the processed sound system. 

As I said, these microphones do not deliver what you hear when standing next to a piano, but as raw-material for further processing lots of character and personality can be added without having to battle audio-physical limitations. I have tested cheap acoustic guitar piezos, Gold Contact mics, Gold suction cup piezos, Scherkler and C-Ducer. I ended up with the Yamahiko/Helpinstill combo. Note that piezo devices often need a good preamp and DI-box with high-ohmi input like for example the Radial ToneBone to deliver a proper signal.

Isolation and microphones

At one point in the project I started to think of the microphone as a gate. The gate width is decided by the microphone type and the placement of it. The gate can be wide open, or it can just have a very narrow opening that lets just a narrow spectre of the sound through. The decision on what parts of the sound this imaginary gate let through depends on what we are using the sound for. Is it for drastic processing or just a slight level increment? How do we want the finished output to sound like? How does the processing setup react to this particular microphone configuration? What does this configuration do to the playing? 

A piezo microphone, depending on quality and signal path, will gate out much sound info with a high amount of isolation. A humbucker even more. A condenser will have a richer sound representation, but larger leaks from other sound sources and more feedback issues. For every new musical setting and room, we work on finding this balance between isolation, levels and sound quality. There are different microphone types that cater different perspectives. The ‘analytic’ and correct cardioid condenser microphones like some distance, but we use them very close to the instrument. In that case they are actually boosting lows and mids of the sound due to the proximity effect. The dynamic microphones (as a middle course) need to be close and the piezo microphones need to be firmly attached to the surface of the instrument in order to capture the physical vibrations of the instrument. 

There is great potential in starting at the first link in the signal chain (the microphone) when we aim for an artificially processed but still warm sound output in the end. Every microphone has a different frequency response and behaviour. Change of microphone and placement can yield subtle or powerful impacts on processing and playing. Also, when moving the microphone around, we might find the perfect spot that has the sound characteristics we need for our use.

Ingar’s method

In a concert situation, the piano sound needs to be amplified and reconstructed to the audience through a PA system, and then blended with my processings and other sounds. Ingar Hunskaar has through his work on my ensembles and other piano players developed a method that maximizes the piano sound and minimizes the leaks and feedback potentials. He moves in close, using a pair of high quality cigar shaped cardioid condenser microphones. The mics are placed in the area with holes on the metal section of the grand piano, mounting the microphone´s head 1–3 mm from the soundboard. The proximity effect may boost up to 16 db. in the low- and mid-range of the sound depending on the microphone, areas which are often the most difficult to bring out in a live situation. The output coming from this setup must undergo a thorough equalizing to become uniform. With this strategy, there won´t be any big issues with other sound sources leaking into the microphones. In this configuration, the microphones are hard panned, one to each side. 

Microphones we use: Shure KSM137, Neumann KM184, DPA 4011 and Sennheiser MKH50. Other small diaphragm cigar shaped condensers with cardioid pattern may also work for this purpose.

The AKG C411 is also suitable. The microphones are securely attached to the piano with two K&M clamps. The small distance between microphones and sound-board demands a steady and sturdy attachment. 

sound distribution

The acoustic sound comes from every nook of the piano. Wherever you place an ear on the instrument it will sound different. The sound of all these spots is what you hear when striking the keyboard.

A sideway to the traditional way of amplifying electronics and piano is to use exciter loudspeakers on the resonant bottom of the piano, and in this way blend the electronic sound and the acoustic sound inside the instrument. This makes the electronic sounds inhabit the same spots as the acoustic sound. This idea worked well in solo and low-volume settings, but not so well when the surroundings got louder. The exciters do not work well with high volumes. This resulted in two main setups for my instrument: The PA-version and the ‘electroacoustic’ version. 

The PA-version is used in bigger rooms, and in ensemble performances. The ‘acoustic’ version is used in rooms designed for acoustic chamber music in solo and duet settings that are not too loud. My main aim in that regard is that the audience shall have a warm and detailed listening experience, and I choose the version mainly based on the size and acoustic attributes of the room, and the nature of the music that is being performed. 

the electroacoustic version

My idea with this version was to have an instrument with acoustic attributes even though the sounds are electronic. This makes the instrument usable in rooms designed for acoustic music, and the audience will hear the details without amplification, meaning that all sounds, electronic and acoustic, come from inside the instrument. The setup with exciters on the resonant bottom, a portable speaker inside the piano and a sub-woofer under the piano came from this idea. The electronic sounds take on acoustic attributes from residing in that resonance case, and the blend of acoustic and electronic sounds is easily obtained. In addition to this I use the same amplifier setup that I use in the PA-version, placed at my left side pointing towards the audience. 

This version turns the piano into a feedback instrument. By placing the microphone and the exciter towards the same surface, the microphone signal is feeding at very low volumes. By tweaking the frequencies on the microphone signal using EQ, ring modulation, filters or frequency modulation, I can use this feedback option to make music, creating dynamic shapes and sweeps between different feedback frequencies. This option can also be used in the PA version, but then I need a much higher volume before the feedback occurs, due to the loudspeaker and microphone not being pointed towards the same surface.

Loudspeakers as instrument

Volume panning and routing through the use of sound cards and mixers, gives me possibilities of sending sound to multiple outputs simultaneously or separated.

The different channels might have different volume and frequency characteristics. In a normal stereo listening situation we are listening to the left and right channel. Furthermore, if we don´t stick to the traditional stereo setup, we can design listening situations for an audience or ourselves with one or more sound outputs that has a specific spatial placement and/or a special sound characteristic. From that point on we have to be aware how this affects the listening situation.

The HyPer(sonal) Piano can send sounds to several outputs. The combination of these outputs creates a different spatial setup than a stereo system, and the fact that these outputs sound very different feature choices on sound quality. The original Ondes Martenot loudspeaker setup and the GRM Acousmonium has been an inspiration for sending the same sound to different outputs with different sound and spatial characteristics ⁶⁵.

I use this strategy both when playing concerts and recording in the studio. The loudspeakers act as processing instruments, and they impose impacts on the musical result. 

Separation of sound categories

In both versions I like to use separate outputs for the prerecorded material and the other processing. With this setup both the technician and I have more control over levels and EQs on the different sound categories. It is especially important to divide percussive sounds and more continuous sound material.

(Marxuino under construction)

lightning mountain

Reflecting on the visual side of my project initiated a search for interagency between music and visuals. I started a process of making visual elements that was a part of my technical setup and that would respond to audio input in various ways. I wanted to take the focus away from me handling buttons, keys and lights and a sea of cables, which is the first visual impression of my instrument.

I wanted something darker, I wanted to hide the electronics and mystify the scenery a little. I wanted to show my interaction with the music, not the interface. I also wanted to show my body movements rather than what my hands were doing. This made me start using a headlamp and later a light installation on my head when I play the Personal Piano material. Now, I hope that there is a discrepancy between music and scenery in my solo concerts. The Hyper– and Personal– contradiction is shining through, giving away more options of readings. This contradiction reminds me of the HAL 9000. 

The project ‘emotion organ’ by Amanda Stegell inspired me in this search. A ‘machine where players can explore the sensational interplay of feeling, seeing, hearing, smelling and motion’ ⁶⁷. I was looking for a link between sound and lights primarily, and I was open for letting the visual aspects interact with my playing. It was important to me that the object(s) could be transported, and that I could mount and control the setup without any help from extra technical personnel or light designers.

I want to be able to present my project full-scale in various rooms, not being too dependent on technical conditions and personnel. The more I can do myself, the better. The visual device Lightning Mountain became an integrated part of the HyPer(sonal) Piano, as if a sound making device could move into the visual domain. It is made in collaboration with the design collective Drap&Design ⁶⁸, who had made an interactive jacket that I liked, taking on colours from the objects that were touched when wearing it. I thought that their approach to interaction design would be right to investigate a dialogue between music and visuals. Sven Håkon Voldum from Drap&Design writes the following about their approach to designing the ‘Lightning Mountain’:

When we started working with Morten Qvenild to develop a new light installation for his stage appearances, our initial idea was to create a wearable piece that would light up based on his musical input. However, as we analyzed his behaviour on stage to find the correct approach for this, it quickly became apparent that a lump of wearable electronics might inhibit his performance more than it would complement it. Therefore we decided to focus on his grand piano instead, modeling a piece that would allow light to flow from it alongside the music he creates with it.

Morten did not simply want a ‘visualizer’ that would light up in a specific manner based on a specific sound input. What he wanted was something that could communicate with as well as play on the highly digitized format of his music. In some ways we considered his requests in the line of a visual sparring partner for his audible input, something that he could ‘duel’ with while on stage.

As the modelling process began, we had a few guidelines to keep in mind; it had to be flexible enough to both complement and comment Morten’s music, it had to be shaped so that it would fit any stage setup, and it had to be limited in physical size so that Morten could actually bring it along for his gigs. 

Lightning Mountain is a black, faceted, shapeable scenery with an inner glow of endless colour variations, that can be draped around Morten’s grand piano. It shows blobs of light in different colours passing across the scenery, with their speed and size depending on what Morten plays or sings. From a little control panel next to him, Qvenild can adjust both responsivity and presets of the light, from relatively calm three-colour-presets making their way comfortably down the length of the installation, through more staccato, fragmented neon-like appearances, or simply a number of white sparks sweeping randomly over the Lightning Mountain.

To achieve this effect, we use RGB LED-strips from Adafruit which are programmed in sequence and controlled with an Arduino Mega. This hardware allows for almost unlimited appearances, since each single LED can be given a specific colour and brightness. Fading them in and out using a unique string of code makes the light appear to flow across the scenery.

Moog Pianobar: A stave that is resting on the wooden board on each side of the piano keyboard, having infrared sensors watching each key. Hooked up with an USB cable to a hardware box. Picks up note and velocity information and sends out midi messages based on this information

Other tests: There are no other similar devices that can be transported and mounted as fast as the Pianobar. You have pianos with fixed midi systems, and there are systems like the TouchKeys by Andrew McPherson, but they are permanent systems and not suited for travelling.

(From a rehearsal with Al Khowarizmis Mekaniske Orkester, July 2016)

The electroacoustic version in duet with Al Khowarizmis Mekaniske Orkester. The UE boom speaker is placed in the resonance case and the genelec-sub is under the instrument. 

Patching

Mic setup:Yamahiko/Helpinstill for processing only, mic Bundle for FX.

CPU 1 stereo:Sent to exciters/UE Boom/Genelec Sub.

Prerecorded tracks CPU 1:Sent to exciters/UE Boom/Genelec Sub.

2880 looper/CPU 2:Sent to exciters/UE Boom/Genelec Sub.

FX pedal section:Two amplifiers pointing towards the audience. 

the PA-version

The PA-version utilizes the PA-system in the concert hall, its mixing console, amplifiers, monitor system and wiring. All my local outputs like amplifiers, computers, noise boxes and voice are sent through that system. I see to that my outputs are mostly finished, leaving it up to the sound engineer to do the final adjustments and levelling. If there are problems sound-wise with some outputs, I will work with the engineer to solve them at my end. (See ”sound control” on page 97)

Tuning

When using a PA and monitoring system, it is very important that the systems are tuned with a graphic EQ before the sound is set. If there are frequency areas that are too loud in the monitors or PA, these areas may accumulate and create an overload of energy in parts of the spectrum. 

Mic setup 

The considerations on isolation, reduction and lo-fi/hi-fi processing has resulted in the following setup for piano processing and amplification. 

1/2: Condenser stereo pair, Ingars Method for PA. 

3:Yamahiko/Helpinstill, for processing.

4:Yamahiko/Helpinstill, for PA.

Note: If I use one Yamahiko only and split it to processing and PA, I will have more accumulation in some frequency areas, because the technician and I are working with the same starting point. 

5/6:Cheap acoustic guitar piezos for sound effects (L/R).

7:AKG C411 attached to brass slide to play on strings.

Patching 

Piano microphones stereo: Sent to PA and monitor fold-back.

CPU 1 stereo:Sent to PA and monitor fold-back.

Prerecorded tracks CPU 1:To PA on separate outs and monitor. 

2880 Looper/CPU 2:Sent to PA and monitor fold-back.

Fx pedal section:Two amplifiers locally, microphone to PA.

Amplifiers

I use two amplifiers to distribute sound from the pedal section. For the A and B chain, I prefer to use a Fender Deluxe 65 Reissue. I have played several of the most common guitar amplifiers with this setup, like the VOX AC30, Fender Twin, Fender HotRod Deluxe, Roland Jazzchorus, Hiwatt custom 50 and Peavey Classic 30. My preference is based on this amp having the largest headroom before feedback, and an even sound output throughout the spectrum. With this setup I need four inputs on two different sections, and the deluxe has this feature. It is also a common amp which is easy for the venues to provide.

For the C chain, I use a Marshall mini amp. This amp is taking away lots of the tone in the low and high area, effectively reducing feedback in the low end and creating a sonic signature that I like for the stuttering-effect chain. I prefer using small amps because they produce sound that are easier to place in a busy music than a full bodied sound from a bigger amplifier. 

In this pass the midi is being recorded:

marxuino

The Marxuino. It consists of a Marxophone, a zither-like US-built instrument with steel hammers on a mechanism giving the string a ‘ping pong ball hits the string and gravity works’ kind of sound. On this instrument, I´ve attached six eBows. Originally I wanted to attach eBows to the grand piano, but there were too many obstacles getting this device work the way I wanted, so I chose to attach the device to the Marxophone and see what happened. The strings are far easier than the Grand to set in motion, and it´s easy to modify the sound with steel pins and other preparation methods on the Marxophone. It can also be amplified and processed with the same modules as the rest of the piano.

I also liked the resemblance and the difference between this instrument and a grand piano, they match each other well sonically. Here is a video of the attempt to use ebows on the grand piano: (from jöK & seasicK #5, 30 May 2013, NMH)

TECHNICAL MANUAL

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From jöK & seasicK #10, 19 January 2015, NMH

From jöK & seasicK #10, 19 January 2015, NMH