Since we already had a substantial documentation of practical experiments from earlier sessions, we decided to not document these sessions with video and written reflections. One could of course say that the sessions are still documented with audio recordings, although I would characterize these different modes of documentation so differently that they require different treatment and impacts the production and the reflection if very different ways. We as musicians are quite used to working in the studio, used to being recorded, and familiar with the scrutiny that follows from this process. Close miking in an optimal recording environment brings out details that you would not necessarily notice otherwise.  The impact of the difference in documentation methods and documentation purpose is the aim of this blog post.

During the practical sessions earlier in this project, we had done what I would call a full video documentation. This consisted of a written session report form (with objectives, listing of participants and takes, reflections during and after the session), a 4-camera video recording of the full session, with reflection interviews after the session, a video digest with highlights of the raw video takes, and a blog post summing up and presenting the results. This recipe was developed as a collaboration between Andreas Bergsland and myself, with Bergsland leading the interviews and taking care of all parts of the video production. A documentation software tool (docmarker) was also developed by me, to aid in marking significant moments in time during sessions, with the purpose of speeding up the process of sifting through the material when editing.

Such an extensive documentation format of course requires substantial resources, but care was taken to try to ensure that the artistic process should not be slowed down by documentation work. We had dedicated personell doing the documentation, for most sessions Bergsland would transparently take care of the video production. The reflective interviews were done after all performing activities were completed, and we also had considerable technical assistance from studio engineers (led by Thomas Henriksen). All this should ensure that any burden of documentation should not impose any hindrance to the workflow of the artistic process being documented.

Still, when we did these final rehearsals in May 2018 without this kind of documentation, I realized that the process and the workflow was much faster. It seems for me like the act of being documented influence the flow of the process in a very significant manner. I relate this to a difference between a pure artistic process, and an artistic research process. I do not propose that what I say here has any form of general validity, but I do recognize it within my own experience from this project and from earlier artistic research projects. It appears to be related to being aware of one’s own intentions on an intellectual level, as opposed to “just doing”. During my doctoral research project “New creative possibilities through improvisational use of compositional techniques, – a new computer instrument for the performing musician” finished in 2008, I also noted this difficulty of doing research on my own process. The being inside while looking from the outside.  The intuitive and the analytic. The subjective and the attempt at being objective. During an earlier session in the crossadaptive project, Bergsland asked me right after a take what I had done during that take. I replied that I did exactly what I said I should (before the take). He then noted that it did not sound as if this was what happened, and I needed to check my settings and realized, yes, … I had actually done some adjustments early in the take “just to make it work”.

After this insight in the May sessions, I have discussed this issue with several collagues. Many seem to agree with me that it would be interesting to try to document the moment when an artistic process ignites, the moment when “it happens”. Many also recognize the inevitable slowdown due to an analytical component being introduced into the inner loop of the process. Others would say that I must be doing something wrong if the act of documentation slows me down so much. The way I experience this, the artistic process has several layers, each with it’s own iteration time. Parts of the process iterates on the time scale of years, where reflective insight and slow learning influence the growth of ideas. Other intermediary layers also exist as onion skin layers before we get to the inner loop that is active during performance. Here, the millisecond interactions between impulses and reactions is at the most sensitive to interruptions and distractions. Any kind of small drag at this level impedes the flow and slows down the loop, sometimes to the extent that it will not work, it stops. As if it was a rotating wheel with a self-preserving spin, delicately balancing between friction and flow. On the longer time scales, documentation does not imede the process, but rather it can enrich it. On those time scales I also recognize the accumulation of material, of background research, technical solutions, philosophical perspectives and so on. It may seem like the gunpower to allow something to ignite is collected during those longer loops, while the spark… well that is a fickle and short-lived entity more vulnerable to analytical impact. In quantum physics there are also (as far as I know) particles too small to be observed, where the act of observation collapses a potentiality. Maybe there is a similar uncertainty principle in artistic research?

Full text available here:
https://escholarship.org/uc/item/0641v1r0

In addition to the researchers presenting, we also had an audience of students from the music technology and the jazz departments, as well as other researchers and teachers from NTNU. We are grateful for the input from the audience to enrich our discussions.

Program:

Thursday 2. November

Practical experiments

Introduction and status. [slides]
Øyvind Brandtsegg

Performance
Maja S.K. Ratkje, Øyvind Brandtsegg, Miller Puckette (standin for Stian Westerhus)

Work methods and session reports. Experiences, insights, reflections.
Trond Engum (with team)[slides],  Bernt Isak Wærstad (with team) [slides]

Instruments and tools

Instrumental affordances, crossadaptivity as instrumental gesture.
Marije Baalman [slides]

Performance
Tone Åse, Carl Haakon Waadeland, Trond Engum

Instrument design and technological developments. [slides]
Sigurd Saue, Victor Lazzarini, Øyvind Brandtsegg

Friday 3. November

Reflection. Aesthetic and philosophical issues

Documentation methods [slides]
Andreas Bergsland

Performance
Bjørnar Habbestad, Gyrid N. Kaldestad, Bernt Isak Wærstad

What does it mean for the performer, for the audience, for the music? How does it change the game?
Solveig Bøe [notes], Simon Emmerson [slides]

Wider use and perspectives

Experiences with Philippe Manoury and Juliana Snapper, thoughts on instrumental control, and a performance
Miller Puckette [PD patches]
(with Øyvind Brandtsegg for a brief liveconvolver performance)

Looking at the music from the mix perspective. Viability of effects as expression. The wider field of automatic mixing and adaptive effects.
Gary Bromham [slides], Josh Reiss [slides]

Outcomes and evaluation. [slides]
Moderator: Øyvind Brandtsegg

Introduction

 During the last several years, the interplay between processed and acoustic sounds has been the focus of research at the department of music technology at NTNU. So far, the project “Cross-adaptive processing as musical intervention” has been mainly concerned with improvised music such as shown with the music of T-EMP. However, through discussions with Øyvind Brandtsegg and several others at NTNU, I have come to find several aspects of their work interesting in the world of written composition, especially mixed music. Frengel (2010) defines this as a form of electroacoustic music which combines a live and/or acoustic performer with electronics. During the last few years I have especially come to be interested in Philippe Manoury’s conception of mixed music which he calls real-time music (la musique du temps réel). This puts emphasis on the idea of real-time versus deferred time which we will come back to later in this article.

The aspect of the cross-adaptive synthesis project which makes the most resonance with mixed music is the idea of adaptive parameters in its most basic form. A parameter X of a sound, influences parameter Y of another sound. To this author, this idea can be directly correlated to Philippe Manoury’s conception of partitions virtuelles which could be translated into English as virtual partition or sheet music. In this article, we will explore some of the links between these concepts, and especially how they can start to influence our compositional practices in mixed music. A few examples of how a few composers have used adaptive parameters will also be given. It is also important to point out that the composers named in this article are far from being the only ones using adaptive parameters, they have done it in either an outstanding and/or very pedagogical way when it comes to the technical aspect of adaptive parameters.

Partitions virtuelles, Relative Values & Absolute Values 

Let us first establish the term virtual partition. Manoury’s (1997) conception comes originally from discussions with the late Pierre Boulez, although it was Manoury that expanded on the concept and extended it through Compositions such as “En écho” (1993-1994), “Neptune” (1991), and “Jupiter” (1987).

Virtual partition is a concept that refers directly to the idea of notation as it is used with traditional acoustic music. If we think of Beethoven’s piano sonatas, the notation tells us which notes to play (ie. C4 followed by E4, etc). It might also contain some tempo marks, dynamic markings and phrasing. Do these parameters form the complete list of parameters that a musician can play? The simple answer is no. These are only basic, and often parameters of relative value. A fortissimo symbol does not mean “play at 90 dB”, the symbol is relative compared to what has come before and what will come after. The absolute parameter that we have in this case, is the notes that are written in the sheet music. An A4 will remain an A4. This duality of seeing absolute against relative is also what has given us the world of interpretation. Paul Lewis will not play the same piece in the exact same way as András Schiff even though they will be playing the exact same notes. This is a vital part of the richness of the classical repertoire: its interpretation.

As Manoury points out, in early electroacoustic music this possibility of interpretation was not possible. The so-called “tyranny of tape” made it rather difficult to have more relative values, and in fact also limited the relative values of the live musicians as their tempo and dynamics had to take into consideration a medium that could not be changed then and there. This aspect of the integration of the electronics is a large field into itself which interests this author very much. Although an exhaustive discussion of this is far beyond the scope of this article, it should be noted that adaptive parameters can be used with most of these integration techniques. However, at this point this author does believe that score following and its real-time relative techniques permit this level of interactivity on a higher plane than say the scene system such as in the piece “Lichtbogen” (1986-1987) by Kaija Saariaho where the electronics are set at specific levels until the next scene starts.

Therefore, the main idea of the virtual partition is to bring interpretation into the use of electronics within mixed music. The first aspect to be able to do so is to work with both relative and absolute variables. How does this relate to adaptive parameters? By using electronics that have several relative values that are influenced either by the electronics themselves, the musician(s) or a combination of both, it becomes possible to bring interpretation to the world of electronics. In other words, the use of adaptive parameters within mixed music can bring more interpretation and fluidity into the genre.

Time & Musical Time

The traditional complaint and/or reason for not implementing adaptive parameters/virtual partitions in mixed music has been its difficulty. The Cross-adaptive synthesis project has proven that complex and beautiful results can be made with adaptive parameters. The flexibility of the electronics can only make the rapport between the musicians and computer a more positive one. Another reason that has often been cited is if the relationships would be clear to the listener. This author feels that this criticism is slightly oblique. The use of adaptive parameters may not always be heard by the audience, but it does influence how much the performer(s) feel connected to the electronics. It is also the basis of being able to create an electronic interpretation. Composers like Philippe Manoury for example, believe that the use of a real-time system with adaptive parameters is the only way to preserve the expressivity while working between electronic and acoustic instruments (Ramstrum, 2006).

Another problem that has often come up when it comes to written music, is how to combine the precise activity of contemporary writing with a computer? Back in the 80’s composers like Philippe Manoury often had to use the help of programmers such as Miller Puckette to come up with novel ideas (which in their case would later lead to the development of MaxMSP and Pure Data). However, the arrival of more stable score followers and recognizers (a distinction brought to the light in Manoury, 1997, p.75-78) has made it possible to think of electronics within musical time (ex. quarter note) instead of absolute time (ex. milliseconds). This also allows a composer to further integrate the interpretation of the electronics directly into the language of the score. One could understand this as the computer being able to translate information from a musical language to a binary language.

To give a simple example, we can assume that the electronic process we are controlling is a playback file on a computer. In the score of our imaginary piece, in 5 measures the solo musician must go from fortissimo to pianissimo and eventually fading out niente (meaning gradually fading to silence). As this is happening, the composer would like the pitch of the playback file to rise, as the amplitude of the musician goes down. In absolute time, one would have to program the number of milliseconds each measure should take and hope that the musician and computer would be in sync. However, by using relative time it is easier, as one can tell the computer that the musician will be playing 5 measures. If we combine this with adaptive parameters, we can directly (or indirectly, if preferred) connect the amplitude of the musician to the pitch of the playback for those 5 measures. This idea, which seemed to be most probably a dream for composers in the 80’s has now become reality  because of programs like Antescofo (described in Cont, 2008) as well as the concept of adaptive parameters which is becoming more commonplace.

The use of microphones and/or other sensors allows us to extract more information from the musician(s) and one can connect these to specific parameters however directly or indirectly one wishes. Øyvind Brandtsegg’s software also allows the extraction of many features of sound(s) and map these to audio processing parameters. In combination with a score follower or other integration methods, this can be an incredibly powerful tool to organize a composition.

Examples from the mixed music repertoire should also be named to give inspiration but also show that it is in fact possible, and already in use. Philippe Manoury has for example used parameters of the live musician(s) to calculate synthesis in many pieces ranging from “Pluton” (1988) to his string quartet “Tensio” (2010). In both pieces, several parameters (such as pitch) are used to create Markov chains and to control synthesis parameters.

Let us look a bit deeper into Manoury’s masterpiece “Jupiter” (1987). Although this piece is an early example of the advanced use of electronics with a computer, the use of adaptive parameters was used actively. For a detailed analysis and explanation of the composition, refer to May (2006). In sections VI and XII the flutist’s playing influences the change of timbre over time. The attack of the flutist’s notes (called note-incipits in May, 2006) control the frequency of the 28 oscillators of the chapo synthesizer – a type of additive synthesis with spectral envelope (Ibid, p. 149) – and its filters. The score also shows these changes in timbre (and pitch) throughout the score (Manoury, 2008, p.32). This is around the 23:06 mark in the recording done my Elizabeth McNutt (2001).

Several other sections’ temporality is also influenced by the performer. For example, in section II, the computer will record short instances of the performer which only be used later in section V and IX in which these excerpts are interpolated and then morphed into tam-tam hits or piano. As May (2006) notes, the flutist’s line directly affects the shape, speed and direction of the interpolations marking a clear relationship between both parts. The sections in which the performer is recorded is clearly marked in the score. In several of the interpolation sections, the electronics take the lead and the performer is told to wait for certain musical elements. The score is also clear about how the interpolations are played and how they are directly related to the performer’s actions.

The composer Hans Tutschku has also used this idea in his series “Still Air” which currently features three compositions (2013, 2014, 2014). In all the pieces, the musicians are to play along to an iPad which might be easier than installing MaxMSP, a soundcard and external microphone for many musicians. The iPad’s built in microphone is used to measure the musicians’ amplitude which varies the amplitude and pitch of the playback files. The exact relationship between the musician and electronics vary throughout the composition. This means that the way the amplitude of the signal modifies the pitch and loudness of the playback part will vary depending on the current event number which are shown in the score (Tutschku, personal communication, October 10, 2017). This use of adaptive parameters is simple and still permits the composer to have a large influence between performer and computer.

A third and final example is “Mahler in Transgress” (2002-2003) by Flo Menezes, especially its restoration done in 2007 by Andre Perrotta which uses Kyma (Perrotta, personal communication, October 6, 2017). Parameters from both pianists are used to control several aspects of the electronics. For example, the sound of one piano could filter the other, or one’s amplitude could affect the spectral profile of the other. Throughout the duration of the composition, the electronics are mainly processing both pianos, as they influence their timbre between each other. This creates a clear relationship between what both performers are playing and the electronics that the audience can hear.

These are only three examples that this author believes shows many different possibilities for the use of adaptive parameters in through-composed music. It is by no means meant to be an exhaustive list, but only as a start to have a common language and understanding on adaptive parameters in mixed music.

A Composition Must Be Like the World or… ?

Gustav Mahler’s (perhaps apocryphal) citation “A symphony must be like the world. It must contain everything” is often popular with composition students. However, this author’s understanding of composition especially in the field of mixed music has led him to believe that a composition should be a world on its own. Its rules should be guided by the poetical meaning of the composition. The rules and ideas that will suit one composition’s electronics and structures, will not necessarily suit another composition. Each composition (its whole of the score and the computer program/electronics) forms its own world.

With this article, this author does not wish to delve into a debate over aesthetics. However, these concepts and possibilities tend to go towards music done in real-time. For years, it was thought that the possibilities of live electronics were limited, but in recent years this has not been the case. The field is still ripe with the possibilities of experimentation within the world of through-composed music.

At this point in time, this author is also experimenting directly with the concept of cross-adaptive synthesis written into through-composed music. I see no reasons as to why it shouldn’t be used both in and out of freely improvised music. We should think of technology and its concepts not within aesthetic boundaries, but as how we can use it for our own purposes.

Bibliography

Cont, A. (2008). “ANTESCOFO: Anticipatory synchronization and control of interactive parameters in computer music.” in Proceedings of international computer music conference (ICMC). Belfast, August 2008.

Frengel, M. (2010). A Multidimensional Approach to Relationships between Live and Non-live Sound Sources in Mixed Works. Organised Sound, 15(2), 96–106. https://doi.org/10.1017/S1355771810000087

Manoury, P. (1997). Les partitions virtuelles in « La note et le son: Écits et entretiens 1981-1998 ». Paris, France: L’Harmattan.

Manoury, P. (1998). Jupiter (score, originally 1987, revised in 1992, 2008). Italy: Universal Music MGB Publishing.

Mcnutt, Elizabeth (2001). Pipewrench: flute & computer (Music recording). Emf Media.

May, Andre (2006). Philippe Manoury’s Jupiter. In Simoni, Mary (Ed.), in Analytical methods of electroacoustic music (pp. 145-186). New York, USA : Routledge Press.

Ramstrum, Momilani (2006). Philippe Manoury’s opera K…. In Simoni, Mary (Ed.), in Analytical methods of electroacoustic music (pp. 239-274). New York, USA : Routledge Press.

Location: NTNU, Studio Olavshallen.

Date: August 28 2017

Participants:
Sissel Vera Pettersen, vocals
Ingrid Lode, vocals
Heidi Skjerve, vocals
Tone Åse, vocals

Øyvind Brandtsegg, processing
Andreas Bergsland, observer and video documentation
Thomas Henriksen, sound engineer
Rune Hoemsnes, sound engineer

We also had the NTNU documentation team (Martin Kristoffersen and Ola Røed) making a separate video recording of the session.

Session objective and focus:
We wanted to try out crossadaptive processing with similar instruments. Until this session, we had usually used it on a combination of two different instruments, leading to very different analysis conditions. The analysis methods reponds a bit differently to each instrument type, and they also each “trigger” the processing in particular manner. It was thought interesting to try some experiments under more “even” conditions. Using four singers and combining them in different duo configurations, we also saw the potential for gleaming personal expressive differences and approaches to the crossadaptive performance situation. This also allowed them to switch roles, i.e. performing under the processing condition where they previously had the modulating role. No attempt was made to exhaustively try every possible combination of roles and effects, we just wanted to try a variety of scenarios possible with the current resources. The situation proved interesting in so many ways, and further exploration of this situation would be neccessary to probe further the research potential herein.
In addition to the analyzer-modulator variant of crossadaptive processing, we also did several takes of live convolution and streaming convolution. This session was the very first performative exploration of streaming convolution.

We used a reverb (Valhalla) on one of the signals, and a custom granular reverb (partikkelverb) on the other. The crossadaptive mappings was first designed so that each of the signals could have a “prolongation” effect (larger size for the reverb, more time smearing for the granular effect). However, after the first take, it seemed that the time smearing for the granular effect was not so clearly perceived as a musical gesture. We then replaced the time smearing parameter of the granular effect with a “graininess” parameter (controlling grain duration). This setting was used for the remaining takes. We used transient density combined with amplitude to control the reverb size, where louder and faster singing would make the reverb shorter (smaller). We used dynamic range to control the time smearing parameter of the granular effect, and used transient density to control the grain size (faster singing makes the grains shorter).

Video digest of the session

Crossadaptive analyzer-modulator takes

Crossadaptive take 1: Heidi/Ingrid
Heidi has a reverb controlled by Ingrids amplitude and transient density
– louder and faster singing makes the reverb shorter
Ingrid has a time smearing effect.
– time is more slowed down when Heidi use a larger dynamic range

Crossadaptive take 2: Heidi/Sissel
Heidi has a reverb controlled by Sissels amplitude and transient density
– louder and faster singing makes the reverb shorter
Sissel has a granular effect.
– the effect is more grainy (shorter grain duration) when Heidi play with a higher transient density (faster)

Crossadaptive take 3: Sissel/Tone
Sissel has a reverb controlled by Tones amplitude and transient density
– louder and faster singing makes the reverb shorter
Tone has a granular effect.
– the effect is more grainy (shorter grain duration) when Sissel play with a higher transient density (faster)

Crossadaptive take 4: Tone/Ingrid
Ingrid has a reverb controlled by Tones amplitude and transient density
– louder and faster singing makes the reverb shorter
Tone has a granular effect.
– the effect is more grainy (shorter grain duration) when Ingrid play with a higher transient density (faster)

Crossadaptive take 5: Tone/Ingrid
Same settings as for take 4

Convolution

Doing live convolution with two singers was thought interesting for the same reasons as listed in the introduction, creating a controlled scenario with two similarly-featured signals. As vocal is in itself one of the richest instruments in terms of signal variation, it was also intersting to explore convolution wwith these instruments. We used the now familiar live convolution techniques, where one of the performers record an impulse response and the other plays through it. In addition, we explored streaming convolution, developed by Victor Lazzarini as part of this project. In streaming convolution, the two signals are treated even more equally that what is the case in live convolution. Streaming convolution simply convolves two circular buffers of a predetermined length, allowing both signals to have the exact same role in relation to the other. It also has a “freeze mode”, where updating of the buffer is suspended, allowing one or the other (or both) of the signals to be kept stationary as a filter for the other. This freezing was controlled by a physical pedal, in the same manner as we use a pedal to control IR sampling with live convolution. In some of the videos one can see the singers raising their hand, as a signal to the other that they are now freezing their filter. When the signal is not frozen (i.e. streaming), there is a practically indeterminate latency in the process as seen from the performer’s perspective. This stems from the fact that the input stream is segmented with respect to the filter length. Any feature recorded into the filter will have a position in the filter dependent on when it was recorded, and the perceived latency between an input impulse and the convolver output of course relies on where in the “impulse response” the most significant energy or transient can be found. The techical latency of the filter is still very low, but the perceived latency depends on the material.

Liveconvolver take 1: Tone/Sissel
Tone records the IR

Liveconvolver take 2: Tone/Sissel
Sissel records the IR

Liveconvolver take 3: Heidi/Sissel
Sissel records the IR

Liveconvolver take 4: Heidi/Sissel
Heidi records the IR

Liveconvolver take 5: Heidi/Ingrid
Heidi records the IR

Streaming Convolution

These are the very first performative explorations of the streaming convolution technique.

Streaming convolution take 1: Heidi/Sissel

Streaming convolution take 2: Heidi/Tone

During this session, we also explored the live convolver in terms of how the audio content in the IR affects the resulting creative options and performative environment for the musician playing through the effect. The liveconvolver takes are presented interspersed with the crossadaptive “feature-modulator” (one could say “proper crossadaptive”) takes. Recording of the impulse response for the convolution was triggered via an external pedal controller during performance, and we let each musician in turn have the role of IR recorder.

Participants:
Jordan Morton: double bass and voice
Miller Puckette: guitar
Steven Leffue: sax
Kyle Motl: double bass
Oeyvind Brandtsegg: crossadaptive mapping design, processing
Andrew Munsie: recording engineer

The music played was mostly free improvisations, but two of the takes with Jordan Morton was performances of her compositions. These were composed in dialogue with the system, during and in between, earlier sessions. She both plays the bass and sings, and wanted to explore how phrasing and shaping of precomposed material could be used to expressively control the timbral modulations of the effects processing.

Jordan Morton: bass and voice.

These pieces are composed by Jordan, and she has composed it with an intention of being performed freely, and shaped according to the situation at performance time, allowing the crossaptive modulations ample room for influence on the sound.

“I confess” (Jordan Morton). Bass and voice.

“Backbeat thing” (Jordan Morton). Bass and voice.

The effects used:
Effects on vocals: Delay, Resonant distorted lowpass
Effects on bass: Reverb, Granular tremolo

The features and the modulator mappings:
(also stating an intended purpose for each mapping)

• Bass spectral flatness, and
• Bass spectral flux: both features giving lesser reverb time on bass

Purpose: When the bass becomes more noisy, it will get less reverb

• Vocal envelope dynamics (dynamic range), and
• Vocal transient density: both features giving lower lowpass filter cutoff frequency on reverb on bass

Purpose: When the vocal becomes more active, the bass reverb will be less pronounced

• Bass transient density: higher cutoff frequency (resonant distorted lowpass filter) on vocal

Purpose: to animate a distorted lo-fi effect on the vocals, according to the activity level on bass

• Vocal mfcc-diff (formant strength, “pressed-ness”): Send level for granular tremolo on bass

Purpose: add animation and drama to the bass when the vocal becomes more energetic

• Bass transient density: lower lowpass filter frequency for the delay on vocal

Purpose: clean up vocal delays when basse becomes more active

• Vocal transient density: shorter delay time for the delay on vocal
• Bass spectral flux: longer delay time for the delay on vocal

Purpose: just for animation/variation

• Vocal dynamic range, and
• Vocal transient density: both features giving less feedback for the delay on vocal

Purpose: clean up vocal delay for better articulation on text

Liveconvolver tracks Jordan/Jordan:

The tracks are improvisations. Here, Jordan’s voice was recorded as the impulse response and she played bass through the voice IR. Since she plays both instruments, this provides a unique approach to the live convolution performance situation.

Liveconvolver take 1: Jordan Morton bass and voice

Liveconvolver take 2: Jordan Morton bass and voice

Jordan Morton and Miller Puckette

Liveconvolver tracks Jordan/Miller:

These tracks was improvised by Jordan Morton (bass) and Miller Puckette (guitar). Each of the musicians was given the role of “impulse response recorder” in turn, while the other then played through the convolver effect.

Improvised liveconvolver performance, Jordan Morton (bass) and Miller Puckette (guitar). Miller records the IR.

Improvised liveconvolver performance, Jordan Morton (bass) and Miller Puckette (guitar). Jordan records the IR.

Discussion on the performance with live convolution, with Jordan Morton and  Miller Puckette.

Miller Puckette and Steven Leffue

These tracks was improvised by Miller Puckette (guitar) and Steven Leffue. The feature-modulator mapping was designed to enable a rich interaction scenario for the performers to explore in their improvisation. The musicians were given only a very brief introduction to the specifities of the mapping before the first take. The intention of this strategy was to create an naturally flowing environment of exploration, with not-too-obvious relationships between instrumental gestures and resulting modulations. After the first take, some more detail of selected elements (one for each musician) of the mapping were repeated for the performers, with the anticipation that these features might be explored more consciously.

Take 1:

Crossadaptive improvisation with Miller Puckette (guitar) and Steven Leffue (sax). Take 1.  Details of the feature-modulator mapping is given below.

Discussion 1 on the crossadaptive performance, with Miller Puckette and Steven Leffue. On the relationship between what you play and how that modulates the effects, on balance of monitoring, and other issues.

The effects used:
Effects on guitar: Spectral delay
Effects on sax: Resonant distorted lowpass, Spectral shift, Reverb

The features and the modulator mappings:
(also stating an intended purpose for each mapping)

• Guitar envelope crest: longer reverb time on sax

Purpose: dynamic guitar playing will make a big room for the sax

• Guitar transient density: higher cutoff frequency for reverb highpass filter and lower cutoff frequency for reverb lowpass filter

Purpose: when guitar is more active, the reverb on sax will be less full (less highs and less lows)

• Guitar transient density (again): downward spectral shift on sax

Purpose: animation and variation

• Guitar spectral flux: higher cutoff frequency (resonant distorted lowpass filter) on sax

Purpose: just for animation and variation. Note that spectral flux (especially on the guitar) will also give high values on single notes in the low register (the lowest octave), in addition to the expected behaviour of giving higher values on more noisy sounds.

• Sax envelope crest: less delay send on guitar

Purpose: more dynamic sax playing will “dry up” the guitar delays, must play long notes to open the sending of guitar to delay

• Sax transient density: longer delay time on guitar. This modulation mapping was also gated by the rms amplitude of the sax (so that it is only active when sax gets loud)

Purpose: load and fast sax will give more distinct repetitions (further apart) on the guitar delay

• Sax pitch: increase spectral delay shaping of the guitar (spectral delay with different delay times for each spectral band)

Purpose: more unnatural (crazier) effect on guitar when sax goes high

• Sax spectral flux: more feedback on guitar delay

Purpose: noisy sax playing will give more distinct repetitions (more repetitions) on the guitar delay

Take 2:

Crossadaptive improvisation with Miller Puckette (guitar) and Steven Leffue (sax). Take 2. The feature-modulator mapping was the same as for take 1.

Discussion 2 on the crossadaptive performance, with Miller Puckette and Steven Leffue. Instructions and intellectualizing the mapping made it harder

Liveconvolver tracks:

Each of the musicians was given the role of “impulse response recorder” in turn, while the other then played through the convolver effect.

Improvised liveconvolver performance, Miller Puckette (guitar) and Steven Leffue (sax). Miller records the IR.

Discussion 1 on playing with the live convolver, with Miller Puckette and Steven Leffue.

Improvised liveconvolver performance, Miller Puckette (guitar) and Steven Leffue (sax). Steven records the IR.

Discussion 2 on playing with the live convolver, with Miller Puckette and Steven Leffue.

Steven Leffue and Kyle Motl

Two different feature-modulator mappings was used, and we present one take of each mapping.  Like the mappings used for Miller/Steven, these were designed to enable a rich interaction scenario for the performers to explore in their improvisation. The musicians were given only a very brief introduction to the specifities of the mapping. The mapping used for the first take closely resembles the mapping for Steven/Miller, with just a few changes to accomodate for the different musical context and how the analysis methods responds to the instruments.

• Bass transient density: shorter reverb time on sax
• The reverb equalization (highpass and lowpass was skipped
• Bass envelope crest: increase send level for granular processing on sax
• Bass rms amplitude: Parametric morph between granular tremolo and granular time stretch on sax

In the first crossadaptive take in this duo, Kyle commented that the amount of delay made it hard to play, that any fast phrases just would turn into a mush. It seemed the choice of effects and the modulations was not optimal, so we tried another configuration of effects (and thus another mapping of features to modulators)

This mapping had earlier been used for duo playing between Kyle (bass) and Øyvind (vocal) on several occations, and it was merely adjusted to accomodate for the different timbral dynamics of the saxophone. In this way, Kyle was familiar with the possibilities of the mapping, but not with the context in which it would be used.
The granular processing done on both instrument was done with the Hadron Particle Synthesizer, which allows a multidimensional parameter navigation through a relatively simple modulation interface (X, Y and 4 expression controllers). The specifics of the actual modulation routing and mapping within Hadron can be described, but it was thought that in the context of the current report, further technical detail would only take away from the clarity of the presentation. Even though the details of the parameter mapping was designed deliberately, at this point in the performative approach to playing with it, we just did no longer pay attention to technical specifics. Rather, the focus was on letting go and trying to experience the timbral changes rather than intellectualizing them.

The effects used:
Effects on sax: Delay, granular processing
Effects on bass: Reverb, granular processing

The features and the modulator mappings:
(also stating an intended purpose for each mapping)

• Sax envelope crest: shorter reverb time on bass
• Sax rms amp: higher cutoff frequency for reverb highpass filter

Purpose: louder sax will make the bass reverb thinner

• Sax transient density: lower cutoff frequency for reverb lowpass filter
• Sax envelope dynamics (dynamic range): higher cutoff frequency for reverb lowpass filter

Purpose: faster sax playing will make the reverb less prominent, but more dynamic playing will enhance it

• Sax spectral flux: Granular processing state morph (Hadron X-axis) on bass
• Sax envelope dynamics: Granular processing state morph (Hadron Y-axis) on bass
• Sax rms amplitude: Granular processing state morph (Hadron Y-axis) on bass

Purpose: animation and variation

• Bass spectral flatness: higher cutoff frequency of the delay feedback path on sax
  Purpose: more noisy bass playing will enhance delayed repetitions
• Bass envelope dynamics: less delay feedback on sax
  Purpose: more dynamic playing will give less repetitions in delay on sax
• Bass pitch: upward spectral shift on sax

Purpose: animation and variation, pulling in same direction (up pitch equals shift up)

• Bass transient density: Granular process expression 1 (Hadron) on sax
• Bass rms amplitude: Granular process expression 2 & 3 (Hadron) on sax
• Bass rhythmic irregularity: Granular process expression 4 (Hadron) on sax
• Bass MFCC diff: Granular processing state morph (Hadron X-axis) on sax
• Bass envelope crest: Granular processing state morph (Hadron Y-axis) on sax

Purpose: multidimensional and rich animation and variation

On the second crossadaptive take between Steven and Kyle, I asked: “Does this hinder interaction or does or make something interesting happen?”
Kyle says it hinders the way they would normally play together. “We can’t go to our normal thing because there’s a third party, the mediation in between us. It is another thing to consider.” Also, the balance between the acoustic sound and the processing is difficult. This is even more difficult when playing with headphones, as the dynamic range and response is different. Sometimes the processing will seem very quiet in relation to the acoustic sound of the instruments, and at other times it will be too loud.
Steven says at one point he started not paying attention to the processing and focused mostly on what Kyle was doing. “Just letting the processing be the reaction to that, not treating it as an equal third party. … Totally paying attention to what the other musician is doing and just keeping up with him, not listening to myself.” This also mirrors the usual options of improvisational listening strategy and focus, of listening to the whole or focusing on specific elements in the resulting sound image.

Longer reflective conversation between Steven Leffule, Kyle Motl and Øyvind Brandtsegg. Done after the crossadaptive feature-modulator takes, touching on some of the problems encountered, but also reflecting on the wider context of different kinds of music accompaniment systems.

Liveconvolver tracks:

Each of the musicians was given the role of “impulse response recorder” in turn, while the other then played through the convolver effect.

Discussion 1 on playing with the live convolver, with Steven Leffue and Kyle Motl.

Discussion 2 on playing with the live convolver, with Steven Leffue and Kyle Motl.

Playing on the sound

One thing that has always inspired me with electric and electronic instruments is how the sound of the instrument can be tweaked and transformed. I know I have many fellow performers with me in saying that changing the sound of the instrument completely changes what you can and will play. This is of course true also with acoustic instrument, but it is even more clear when you can keep the physical interface identical but change the sonic outcome drastically. The comparision becomes more clear, since the performative actions, the physical interaction with the instrument does not need to change significantly. Still, when the sound of the instrument changes, even the physical gestures to produce it changes and so also what you will play.  There is a connection and an identification between performer and sound, the current sound of the instrument. This als oextends to the amplification system and to the room where the sound comes out. Performers who has a lot of experience playing on big PA systems know the difference between “just” playing your instrument and playing the instrument through the sound system and in the room.

Automation in instruments

In this context, I have also mused on the subject of how much an instrument ‘does for you’. I mean, automatically, for example “smart” computer instruments that will give back (in some sense) more than you put in. Also, in terms of “creative” packages like Garageband and also much of what comes with programs like Ableton Live, where we can shuffle around templates of stuff made by others, like Photoshop beautifying filters for music. This description is not intended to paint a bleak picture of the future of creativity, but indeed is something to be aware of. In the context of our current discussion it is relevant because of the relation between input and output of the instrument; Garageband and Live, as instruments, will transform your input significantly according to their affordances. The concept is not necessarily limited to computer instruments either, as all instruments add ‘something’ that the performer could not have done by himself (without the external instrument). Also, as an example many are familiar with: playing through a delay effect: Creating beautiful rhythmic textures out of a simple input, where there may be a fine line between the moment you are playing the instrument, and all of a sudden the instrument is playing you, and all you can do is try to keep up. The devil, as they say, is in the delays!

Flow and groove

There is also a common concept among musicians, when the music flows so easily as if the instrument is playing itself. Being in the groove, in flow,  transcendent, totally in the moment, or other descriptions may apply.  One might argue that this phenomenon is also result of training, muscle memory, gut reaction, instinct. These are in some ways automatic processes. Any fast human reaction relies in some aspect on a learned response, processing a truly unexpected event takes several hundred milliseconds. Even if it is not automated to the same degree as a delay effect, we can say that there is not a clean division between automated and conteplated responses. We could probably delve deep into physchology to investigate this matter in detail, but for our current purposes it is sufficient to say automation is there to some degree at this level of human performance as well as in the instrument itself.

Another aspect of automation (if we in automation can include external events that triggers actions that would not have happened otherwise), or of “falling into the beat” is the synchronizing action when playing in rhythm with another performer. This has some aspects of similarity to the situation when “being played” by the delay effect. The delay processor has even more of a “chasing” effect since it will always continue, responding to every new event, non stop. Playing with another performer does not have that self continuing perpetual motion, but in some cases, the resulting groove might have.

Adaptive, in what way?

So when performing in a crossadaptive situation, what attitude could or should we attain towards the instrument and the processes therein? Should the musicians just focus on the acoustic sound, and play together more or less as usual, letting the processing unfold in its own right? From a traditionally trained performer perspective, one could expect the processing to adapt to the music that is happening, adjusting itself to create something that “works”. However, this is not the only way it could work, and perhaps not the mode that will produce the most interesting results. Another approach is to listen closely to what comes out of the processing. Perhaps to the degree that we disregard the direct sound of the (acoustic) instrument, and just focus on how the processing responds to the different performative gestures. In this mode, the performer would continually adjust to the combined system of acoustic instrument, processing, interaction with the other musician, signal interaction between the two instruments, also including any contribution from the amplification system and the ambience (e.g. playing on headphones or on a P.A). This is hard for many performers, because the complete instrument system is bigger, has more complex interactions, and sometimes has a delay from an action occurs to the system responds (might be a musical and desirable delay, or a technical artifact), plainly a larger distance to all the “moving parts” of the machinery that enables the transformation of a musical intent to a sounding result. In short, we could describe it as having a lower control intimacy. There is also of course a question of the willingness of the performer to set himself in a position where all these extra factors are allowed to count, as it will naturally render most of us in a position where we again are amateurs, not knowing how the instrument works. For many performers this is not immediately attractive. Then again, it is an opportunity to find something new and to be forced to abandon regular habits.

One aspect that I haven’t seen discussed so much is the instrumental scope of the performer. As described above, the performer may choose to focus on the acoustic and physical device that was traditionally called the instrument, and operate this with proficiency to create coherent musical statements. On the other hand, the performer may take into account the whole system (where does that end?, is it even contained in the room in which we perform the music?) of sound generation and transformation. Many expressive options and possibilities lies within the larger system, and the position of the listener/audience also oftentimes lies somewhere in the bigger space of this combined system. These reflections of course apply just as much to any performance on a PA system, or in a recording studio, but I’d venture to say they are crystallized even more clearly in the context of the crossadaptive performance.

Intellectualize the mapping?

To what degree should the performers know and care about the details of the crossadaptive modulation mappings? Would it make sense to explore the system without knowing the mapping? Just play. It is an attractive approach for many, as any musical performance situation in any case is complex with many unknown factors, so why not just throw in these ones too? This can of course be done, and some of our experiments in San Diego has been following this line of investigation (me and Kyle played this way with complex mappings, and the Studio A session between Steven an Kyle leaned towards this approach). The rationale for doing so is that with complex crossadaptive mappings, the intellectual load of just remembering all connections can override any impulsive musical incentive. Now, after doing this on some occasions, I begin to see that as a general method perhaps this is not the best way to do it. The system’s response to a performative action is in many cases so complex and relates to so many variables, that it is very hard to figure out “just by playing”. Some sort of training, or explorative process to familiarize the performer with the new expressive dimensions is needed in most cases. With complex mappings, this will be a time consuming process. Just listing and intellectualizing the mappings does not work for making them available as expressive dimensions during performance. This may be blindingly obvious after the fact, but it is indeed a point worth mentioning. Familiarization with the expressive potential takes time, and is necessary in order to exploit it. We’ve seen some very clear pedagogical approaches in some of the Trondheim sessions, and these take on the challenge of getting to know the full instrument in a step by step manner. We’ve also seen some very fruitful explorative approaches to performance in some of the Oslo sessions. Similarly, when Miller Puckette in our sessions in San Diego chooses to listen mainly to the processing (not to the direct sound of his instrument, and not to the direct sound of his fellow musician’s instrument, but to the combined result), he actively explores the farthest reaches of the space constituted by the instrumental system as a whole. Miller’s approach can work even if all the separate dimensions has not been charted and familiarized separately, basically because he focus almost exclusively on those aspects of the combined system output. As often happens in conversations with Miller, he captures the complexity and the essence of the situation in clear statements:

“The key ingredients of phrasing is time and effort.”

What about the analytical listener?

In our current project we don’t include any proper research on how this music is experienced by a listener. Still, we as performers and designers/composers are also experiencing the music as listeners, and we cannot avoid wondering how (or if) these new dimensions of expression affects the perception of the music “from the outside”. The different presentations and workshops of the project affords opportunities to hear how outside listeners perceive it.  One recent and interesting such opportunity came when I was asked to present something for Katharina Rosenbergers Composition Analysis class at UCSD. The group comprised of graduate composition students, critical and highly reflective listeners, and in the context of this class especially aimed their listening towards analysis.  What is in there ? How does it work musically?  What is this composition? Where is the composing? In the discussions with this class, I got to ask them if they perceived it as important for the listener to know and understand the crossadaptive modulation mappings. Do they need to learn the intricacies of the interaction and the processing in the same pedagogical manner? The output from this class was quite clear on the subject:

It is the things they make the performers do that is important

In one way, we could understand it as a modernist stance that if it is in there, it will be heard and thus it matters. We could also understand it to mean that the changes in the interaction, the thing that performers will do differently in this setting is what is the most interesting. When we hear surprise (in the performer), and a subsequent change of direction, we can follow that musically without knowing about the exact details that led to the surprise.

“Phantom Station”, The Loft, SD

This group is based on Butch Morris’ conduction language for improvisation, and the performance typically requires a specific action (specific although it is free and open) to happen on cue from the conductor. I was invited into this ensemble and encouraged to use whatever part of my instrumentarium that I might see fit. Since I had just finished the liveconvoolver plugin, I wanted to try that out. I also figured my live processing techniques would fit easily, in case the liveconvolver did not work so well. Both the live processing and the live convolution instruments was in practice less than optimal for this kind of ensemble playing. Even though the instrumental response can be fast (low latency), the way I normally use these instruments is not for making a musical statements quickly for one second and then suddenly stop again. This leads me to reflect on a quality measure I haven’t really thought of before. For lack of a better word, let’s call it reactive inertia: the possibility to completely change direction on the basis of some external and unexpected signal. This is something else than the audio latency (of the audio processing) and also something else than the user interface latency (like for example, the time it takes the performer to figure out which button to turn to achieve a desired effect). I think it has to do with the sound production process, for example how some effects take time to build up before they are heard as a distinct musical statement, and also the inertia due to interaction between humans and also the signal chain of sound production pre effects (say if you live sample or live process someone, need to get a sample, or need to get some exciter signal). For live interaction instruments, the reactive inertia is then goverened by the time it takes two performers to react to the external stimuli, and their combined efforts to be turned into sound by the technology involved. Much like what an old man once told me at Ocean Beach:

“There’s two things that needs to be ready for you to catch a wave
– You, …and the wave”.

We can of course prepare for sudden shifts in the music, and construct instruments that will be able to produce sudden shifts and fast reactions. Still, the reaction to a completely unexpected or unfamiliar stimuli will be slower than optimal. An acoustic instrument has less of these limitations. For this reason, I switched to using the Marimba Lumina for the remaning two concerts with Phantom Station, to be able to shape immediate short musical statements with more ease.

“Creatures”, The Loft, SD

Creatures is the duo of Jordan Morton (double bass, vocals) and Kai Basanta (drums). I had the pleasure of sitting in with them, making it a trio for this concert at The Loft. Creatures have some composed material in the form of songs, and combine this with long improvised stretches. For this concert I got to explore the liveconvolver quite a bit, in addition to the regular live processing and Marimba Lumina. The convolver was used with input from piezo pickups on the drums, convolving with IR live recorded from vocals. Piezo pickups can be very “impulse-like”, especially when used on percussive instruments. The pickups’ response have a generous amount of high frequencies, and a very high dynamic range. Due to the peaky impulse-like nature of the signal, it drives the convolution almost like a sample playback trigger, creating delay patterns on the input sound. Still the convolver output can become sustained and dense, when there is high activity on the triggering input. In the live mix, the result sounds somewhat similar to infinite reverb or “freeze” effects (using a trigger to capture a timbral snippet and holding that sound as long as the trigger is enabled). Here, the capture would be the IR recording, and the trigger to create and sustain the output is the activity on the piezo pickup. The causality and performer interface is very different than that of a freeze effect, but listening to it from the outside, the result is similar. These expressive limitations can be circumvented by changing the miking technique, and working in a more directed way as to what sounds goes into the convolver. Due to the relatively few control parameters, the main thing deciding how the convolver sounds is the input signals. The term causality in this context was used by Miller Puckette when talking about the relationship between performative actions and instrumental reactions.

Creatures at The Loft. A liveconvolver example can be found at 29:00 to 34:00 with Vocal IR, and briefly around 35:30 with IR from double bass.

“Nordeson/Brandtsegg duo”, Bread & Salt, SD

Duo configuration, where Kjell plays drums/perc and vibraphone, and I did live convolution, live processing and Marimba Lumina. My techniques was much like what I used with Creatures. The live convolver setup was also similar, with IR being live sampled from my vocals and the convolver being triggered by piezo pickups on Kjell’s drums. I had the opportunity to work over a longer period of time preparing for this concert together with Kjell. Because if this, we managed to develop a somewhat more nuanced utilization of the convolver techniques. Still, in the live performance situation on a PA, the technical situation made it a bit more difficult to utilize the fine grained control over the process and I felt the sounding result was similar in function to what I did together with Creatures. It works well like this, but there is potential for getting a lot more variation out of this technique.

We used a quadrophonic PA setup for this concert. Due to an error with the front-of-house patching, only 2 of the 4 lines from my electronics was recorded. Due to this fact, the mix is somewhat off balance. The recording also lacks first part of the concert, starting some 25 minutes into it.

“The Gringo and the Desert”, Coaxial, LA

In this duo Kyle Motl plays double bass and I do vocals, live convolution, live processing, and also crossadaptive processing. I did not use the Marimba Lumina in this setting, so some more focus was allowed for the processing. In terms of crossadaptive processing, the utilization of the techniques is a bit more developed in this configuration. We’ve had the opportunity to work over several months, with dedicated rehearsal sessions focusing on separate aspects of the techniques we wanted to explore. As it happpened during the concert, we played one long set and the different techniques was enabled as needed. Parameters that was manually controlled in parts of the set, was then delegated to crossadaptive modulations in other parts of the set. The live convolver was used freely as one out of several active live processing modules/voices. The liveconvolver with vocal IR can be heard for example from 16:25 to 20:10. Here, the IR is recorded from vocals, and the process acts as a vocal “shade” or “pad”, creating long sustained sheets of vocal sound triggeered by the double bass. Then, liveconvolver with bass IR from 20:10 to 23:15, where we switch on to full crossadaptive modulation until the end of the set. We used a complex mapping designed to respond to a variety of expressive changes. Our attitude/approach as performers was not to intellectually focus on controlling specific dimensions but to allow the adaptive processing to naturally follow whatever happened in the music.

Session with Steven Leffue (Apr 28th, May 5th)

I did two rehearsal sessions together with Steven Leffue in April, as preparation for the UCSD Studio A session in May. We worked both on crossadaptive modulations and on live convolution. Especially interesting with Steven is his own use of adaptive and crossadaptive techniques. He has developed a setup in PD, where he tracks transient density and amplitude envelope over different time windows, and also uses standard deviation of transient density within these windows. The windowing and statistics he use can act somewhat like a feature we have also discussed in our crossadaptive project: a method of making an analysis “in relation to the normal level” for a given feature. Thus, a way to track relative change. Steven’s Master thesis “Musical Considerations in Interactive Design for Performance” relates to this and other issues of adaptive live performance. Notable is also his ICMC paper “AIIS: An Intelligent Improvisational System”. His music can be heard at http://www.stevenleffue.com/music.html, where the adaptive electronics is featured in “A theory of harmony” and “Futures”.
Our first session was mainly devoted to testing and calibrating the analysis methods towards use on the saxophone. In very broad terms, we notice that the different analysis streams now seem to work relatively similar on different instruments. The main diffferences are related to extraction of tone/noise balance, general brightness, timbral “pressedness” (weight of formants), and to some extent in transient detection and pitch tracking. The reason why the analysis methods now appear more robust is partly due to refinements in their implementation, and partly due to (more) experience in using them as modulators. Listening, experimentation, tweaking, and plainly just a lot of spending-time-with-them, have made for a more intuitive understanding of how each analysis dimension relates to an audio signal.
The second session was spent exploring live convolution between Sax and Vocals. Of particular interest here is the comments from Steven regarding the performative roles of IR recording vs playing the convolver. Steven states quite strongly that the one recording the IR has the most influence over the resulting music. This impression is consistent when he records the IR (and I sing through it), and when I record the IR and he plays through it. This may be caused by several things, but of special interest is that it is diametrically opposed to what many other performers have stated. Both Kyle, Jordan and Kjell in our initial sessions, voiced a higher performative intimacy, a closer connection to the output when playing through the IR. Maybe Steven is more concerned with the resulting timbre (including processed sound) than the physical control mechanism, as he routinely designs and performs with his own interactive electronics rig. Of course all musician care about the sound, but perhaps there is a difference of approach on just how to get there. With the liveconvolver we put the performers in an unfamiliar situation, and the differences in approach might just show different methods of problems solving to gain control over this situation. What I’m trying to investigate is how the liveconvolver technique works performatively, and in this, the performer’s personal and musical traits plays into the situation quite strongly. Again, we can only observe single occurences and try to extract things that might work well. There is no conclusions to be drawn on a general basis as to what works and what does not, and neither can we conclude what is the nature of this situation and this tool. One way of looking at it (I’m still just guessing) is that Steven treats the convolver as *the environment* in which music can be made. The changes to the environment determines what can be played and how that will sound, and thus, the one recording the IR controls the environment and subsequently controls the most important factor in determining the music.
In this session, we also experimented a bit with transposed and revesed IR, this being some of the parametric modifications we can make to the IR with our liveconvolver technique. Transposing can be interesting, but also quite difficult to use musically. Transposing in octave intervals can work nicely, as it will act just as much as a timbral colouring without changing pitch class. A fun fact about reversed IR as used by Steven: If he played in the style of Charlie Parker and we reversed the IR, it would sound like Evan Parker. Then, if he played like Evan Parker and we reversed the IR, it would still sound like Evan Parker. One could say this puts Evan Parker at the top of the convolution-evolutionary-saxophone tree….

Session with Miller Puckette, May 8th

The session was intended as “calibration run”, to see how the analysis methods responded to Miller’s guitar. This as a preparation for the upcoming bigger session in UCSD Studio A. The main objective was to determine which analysis features would work best as expressive dimensions, find the appropriate ranges, and start looking at potentially useful mappings. After this, we went on to explore the liveconvolver with vocals and guitar as the input signals. Due to the “calibration run” mode of approach, the session was not videotaped. Our comments and discussion was only dimly picked up by the microphones used for processing. Here’s a transcription of some of Millers initial comments on playing with the convolver:

“It is interesting, that …you can control aspects of it but never really control the thing. The person who’s doing the recording is a little bit less on the hook. Because there’s always more of a delay between when you make something and when you hear it coming out [when recording the IR]. The person who is triggering the result is really much more exposed, because that person is in control of the timing. Even though the other person is of course in control of the sonic material and the interior rhythms that happen.”

Since the liveconvolver has been developed and investigated as part of the research on crossadaptive techniques, I had slipped into the habit of calling it a crossadaptive technique. In discussion with Miller, he pointed out that the liveconvolver is not really *crossadaptive* as such. BUT it involves some of the same performative challenges, namely playing something that is not played solely for the purpose of it’s own musical value. The performers will sometimes need to play something that will affect the sound of the other musician in some way. One of the challenges is how to incorporate that thing into the musical narrative, taking care of how it sounds in itself, and exactly how it will affect the other performer’s sound. Playing with liveconvolver has this performative challenge, as has the regular crossadaptive modulation. One thing the live convolver does not have is the reciprocal/two-way modulation, it is more of a one-way process. The recent Oslo session on liveconvolution used two liveconvolvers simultaneously to re-introduce the two-way reciprocal dependency.

The focus for this session was to work with the new live convolver in Ableton Live

Setup – getting to know the Convolver

We worked in duo configurations – flute/guitar, guitar/vocal and vocal/flute

We started with spending some time exploring and understanding the controls. Our first setup was guitar/flute and we chose to start convolving in auto mode. We knew both from experience with convolution in general and from previous live convolver session reports, that sustained and percussive sounds would yield very different results and we therefore started with combinations: percussive sounds (flute) with sustained (guitar). While this made it quite clear how the convolver worked, the output was less than impressive. Next step was to switch the inputs while preserving the playing technique. Still everything seemed to sound somewhat delayed with ringing overtones. It was suggested to add in some dry signal to produce more aesthetically pleasing sounds, but at this point we decided to only listen to the wet signal, as the main goal was to explore and understand the ways of the convolver.

Sonics of convolution

We continued the process by switching from auto to triggered mode where the flute had the role of the IR to make the convolver a bit more responsive. This produced a few nice moments, but the overall result was still quite “mushy”. We explored reducing the IR size and working with IR pitch to explore ways of getting other sound qualities.

We subsequently switched from flute to vocal, working with guitar and voice in trigger mode where voice was used as IR. We also decided to add in some dry sound from the vocal, since the listeners (Mats and Bjørnar) found it much more interesting when we could hear the relation between the IR and the convolved signal. Since the convolver still felt quite slow in response and muddy in sound, we also tried out very short IR size in auto mode

As shown in this video, we tried to find out how the vocal sound could affect the processing of the flute. The experience was either that the got some strange reverberation or some strange eq- we thought that the sound that came out of the processed flute was not so interesting as we had hoped!

Trying again – doesn’t feel like we get any interesting sounds. Is it the vocal input that doesn’t give the flute sound anything to work with? The flute sound we get out is still a bit harsh in the higher frequencies and with a bit of strange reverberation.

Critical listening

All these initial tests were more or less didactic, in that we chose fixed materials (such as percussive versus sustained) in order to emphasise the effect of the convolver. After three short sessions this became a limitation that hindered improvisational flow and phrasing. Especially in the flute/vocal session this was an issue. Too often, the sonic result of the convolution was less than intriguing. We discussed whether the live convolver would lend itself more easily to composed situations, as the necessity of carefully selecting material types that would convolve in an interesting way rendered it less usable in improvised settings. We decided to add more control to the setup in order to remedy this problem.

Adding control

Gyrid commented that she felt the system was quite limiting, especially when you are used to control live processing yourself. To remedy this, we started adding parameter controls for the performers. In a session with vocal and guitar, we added a expression pedal for Bernt Isak (guitar) to control the IR size. This was the first time we had the experience of a responsive system that made musical sense.

Revised setup

After some frustrating, and from our perception, failed attempts at getting interesting musical results, we decided to revise our setup. After some discussion we came to the conclusion that our existing setup, using one live convolver, was cross adaptive in the signal control domain, but didn’t feel cross adaptive in the musical domain. Therefore we decided to add another crossing by using two live convolvers, where each instrument had the role as IR in one convolver and as input in the other convolver. We also decided to set one convolver to auto mode and the other to trigger mode for better separation and more variation in musical output.

Guitar

• Fader 1: IR size
• Fader 2: IR auto update-rate
• Fader 3: Dry sound
• Expression pedal 1: IR pitch
• Toggle switch: switch inputs

Vocal

• Fader 1: IR size
• Fader 2: IR pitch

Flute

• Fader 1: IR size
• Fader 2: IR pitch
• Fader 3: Dry sound

Convolvers were placed on return tracks – both panned slightly to the sides to easier distinguish the two convolvers, while also adding some stereo width.

Sound excerpt 1- flute & vocal, using two convolvers:

Bjørnar has the same setup as Bernt Isak. Better experience. It could change the way the convolver use the signal if we use different microphones – maybe one inside and one outside the flute.

It’s quite apparent to us that using sustained sounds doesn’t work very well. It seems to us that the effect just makes the flute sound less interesting, it somehow reduces the bandwidth, amplifies the resonant frequencies  or just make some strange phasing. The soundscape is changing and gets more interesting when we shift to more percussive and distorted sound qualities. Could it be an idea to make a it possible to extract only the distorted parts of a sound input?

Sound excerpt 2- guitar & vocal, using two convolvers:

Session with guitar and vocal where we control the IR size, IR pitch and IR rate.

Gyrid has the input signal in trigger mode and can control IR size and pitch with faders. Bernt Isak  has the input signal in auto mode –  and can control amount of dry signal,  IR size and rate with faders and  pitch with an expression pedal. Very positive experience to use two convolvers! Even though one convolver is cross adaptive in the sense that it uses two signals, it didn’t feel cross adaptive musically, but more like a traditional live processing setup. We also found that having one convolver in trigger mode and one in auto mode was a good way of adding movement and variation in the music as one convolver would keep a more steady “timing”, while the other one can be completely free. It also seems essential to have the possibility to control the dry signal – hearing the dry signal makes the music more three dimensional.

Sound excerpt 3- flute & guitar, using two convolvers:

Session with guitar and flute – Bjørnar has the same setup as Gyrid, but with added control for amount of dry sound. Same issue with flute microphone as above

The experience is very different with flute and vocals and guitar and vocals, this has mainly to do with the way the instruments are played. The guitar has a very distinct attack and it is very clear when the timbral character change. Flute and vocals have a more similar frequency response and the result get less interesting. Adding more effects to the guitar (distortion + tremolo) makes a huge difference- but also the fact that percussive sounds from vocal gives the most interesting musical output.

Overall session reflections 

Choice of instrument combinations is crucial for live convolving to be controllable and produce artistically interesting results. We also noted that there is a difference between signal cross adaptiveness and musical cross adaptiveness. From our experience, double live convolving is needed to produce a similar feel of musical cross adaptiveness as we’ve experienced from the previous signal processing cross adaptive sessions:

• http://crossadaptive.hf.ntnu.no/index.php/2016/12/12/oslo-first-session-october-18-2016/
• http://crossadaptive.hf.ntnu.no/index.php/2017/06/07/second-session-at-norwegian-academy-of-music-oslo-january-13-and-19-2017/

Ideas for further development

Some ideas for adding more control – the possibility to switch between auto mode and trigger mode could be interesting. It would also be useful with a visual indicator for the IR trigger mode for easier tuning of the trigger settings.

Bjørnar suggested to combine the live convolver with the analyzer/midimapper in order to only convolve when there is a minimum of noise and/or transients available. E.g. linking spectral crest to a wet/dry parameter in the convolution or splitting the audio signal based on spectral conditions.

It could perhaps also yield some interesting result to add some spectral processing to reduce the fundamental frequency component (similar to Øyvind Brandtseggs feedback tools) for instruments that have a very strong fundamental (like flute).