Second session at Norwegian Academy of Music (Oslo) – January 13. and 19., 2017

Participants: Bjørnar Habbestad (flute), Bernt Isak Wærstad (guitar), Gyrid Nordal Kaldestad (voice)

The focus for this session was to play with, fine tune and work further on the mappings we sat up during the last session at NMH in November. Due to practical reasons, we had to split the session into two half days on 13th and 19th of January

13 th of January 2017

We started by analysing 4 different musical gestures for the guitar, which was skipped due to time constraints during the last session. During this analysis we found the need to specify the spread of the analysis results in addition to the region. This way we could differentiate the analysis results in terms of stability and conclusiveness. We decided to analyse the flute and vocal again to add the new parameters.

19 th of January 2017

After the analysis was done, we started working on a mapping scheme which involved all 3 instruments, so that we could play in a trio setup. The mappings between flute and vocal where the same as in the November session

The analyser was still run in Reaper, but all routing, effects chain and mapping (MIDIator) was now done in Live. Because of software instability (the old Reaper projects from November wouldn’t open) and change of DAW from Reaper to Live, meant that we had to set up and tune everything from scratch.

Sound examples with comments and immediate reflections

1. Guitar & Vocal – First duo test, not ideal, forgot to mute analyser.

2. Guitar & Vocal retake – Listened back on speakers after recording. Nice sounding. Promising.

Reflection: There seems to be some elements missing, in a good way, meaning that there is space left for things to happen in the trio format. There is still need for fine-tuning of the relationship between guitar and vocal. This scenario stems from the mapping being done mainly with the trio format in mind.

3. Vocals & flute – Listened back on speakers after recording.

Reflections: dynamic soundscape, quite diverse results, some of the same situations as with take 2, the sounds feel complementary to something else. Effect tuning: more subtle ring mod (good!) compared to last session, the filter on vocals is a bit too heavy-handed. Should we flip the vocal filter? This could prevent filtering and reverb taking place simultaneously. Concern: is the guitar/vocal relationship weaker compared to vocal/flute? Another idea comes up – should we look at connecting gates or bypasses in order to create dynamic transitions between dry and processed signals?

4.Flute & Guitar

Reflections: both the flute ring mod and git delay are a bit on the heavy side, not responsive enough. Interesting how the effect transformations affect material choices when improvising.

5.Trio

Comments and reflections after the recording session

It is interesting to be in a situation where you, as you play, are having multi-layered focuses- playing, listening, thinking of how you affect the processing of your fellow musicians and how your sound is affected and trying to make something worth listening to. Of course we are now in an “etyde- mode”, but still striving for the goal, great output!

It seems to be a bug in the analyser tool when it comes to being consistent. Sometimes some parameters fall out. We experienced that it seems to be a good idea to run the analyse a couple of times for each sound to get the most precise result.

Seminar on instrument design, software, control

Online seminar March 21

Trond Engum and Sigurd Saue (Trondheim)
Bernt Isak Wærstad (Oslo)
Marije Baalman (Amsterdam)
Joshua Reiss (London)
Victor Lazzarini (Maynooth)
Øyvind Brandtsegg (San Diego)

Instrument design, software, control

We now have some tools that allow practical experimentation, and we’ve had the chance to use them in some sessions. We have some experience as to what they solve and don’t solve, how simple (or not) they are to use. We know that they are not completely stable on all platforms, there are some “snags” on initialization and/or termination that give different problems for different platforms. Still, in general, we have just enough to evaluate the design in terms of instrument building, software architechture, interfacing and control.

We have identified two distinct modes of working crossadaptively: The Analyzer-Modulator workflow, and a Direct-Cross-Synthesis workflow. The Analyzer-Modulator method is comprised of extracting features, and arbitrarily mapping these features as modulators to any effect parameter. The Direct-Cross-Synthesis method is comprised by a much closer interaction directly on the two audio signals, for example as seen with the liveconvolver and or different forms of adaptive resonators. These two methods give very different ways of approaching the crossadaptive interplay, with the direct-cross-synthesis method being perceived as closer to the signal , and as such, in many ways closer to each other for the two performers. The Analyzer-Modulator approach allows arbitrary mappings, and this is both a strngth and a weakness. It is powerful by allowing any mapping, but it is harder to find mappings that are musically and performatively engaging. At least this can be true when a mapping is used without modification over a longer time span. As a further extension, an even more distanced manner of crossadaptive interplay was recently suggested by Lukas Ligeti (UC Irvine, following Brandtsegg’s presentation of our project there in January). Ligeti would like to investigate crossadaptive modulation on MIDI signals between performers. The mapping and processing options for event-based signals like MIDI would have even more degrees of freedom than what we achieve with the Analyzer-Modulator approach, and it would have an even greater degree of “remoteness” or “disconnectedness”. For Ligeti, one of the interesting things is the diconnectedness and how it affects our playing. In perspective, we start to see some different viewing angles on how crossadaptivity can be implemented and how it can influence communication and performance.

In this meeting we also discussed problems of the current tools, mostly concerned with the tools of the Analyzer-Moduator method, as that is where we have experienced the most obvious technical hindrance for effecttive exploration. One particular problem is the use of MIDI controller data as our output. Even though it gives great freedom in modulator destinations , it is not straightforward for a system operator to keep track of which controller numbers are actively used and what destinations they correspond to. Initial investigations of using OSC in the final interfacing to the DAW have been done by Brandtsegg, and the current status of many DAWs seems to allow “auto-learning” of OSC addresses based on touching controls of the modulator destination within the DAW. a two-way communication between the DAW and ouurr mapping module should be within reach and would immensely simplify that part of our tools.
We also discussed the selection of features extracted by the Analyzer, whic ones are more actively used, if any could be removed and/or if any could be refined.

Initial comments

Each of the participants was invited to give their initial comments on these issues. Victor suggests we could rationalize the tools a bit, simplify, and get rid of the Python dependency (which has caused some stability and compatibility issues). This should be done without loosing flexibility and usability. Perhaps a turn towards the originally planned direction of reying basically on Csound for analysis instead of external libraries. Bernt has had some meetings with Trond recently and they have some common views. For them it is imperative to be able to use Ableton Live for the audio processing, as the creative work during sessions is really only possible using tools they are familiar with. Finding solutions to aesthetic problems that may arise require quick turnarounds, and for this to be viable, familiar processing tools.  There have been some issues related to stability in Live, which have sometimes significantly slowed down or straight out hindered an effective workflow. Trond appreciates the graphical display of signals, as it helps it teaching performers how the analysis responds to different playing techniques.

Modularity

Bernt also mentions the use of very simple scaled-down experiments directly in Max, done quickly with students. It would be relatively easy to make simple patches that combines analysis of one (or a few) features with a small number of modulator parameters. Josh and Marije also mentions modularity and scaling down as measures to clean up the tools. Sigurd has some other perspectives on this, as it also relates to what kind of flexibility we might want and need, how much free experimentation with features, mappings and desintations is needed, and also to consider if we are making the tools for an end user or for the research personell within the project. Oeyvind also mentions some arguments that directly opposes a modular structure, both in terms of the number of separate plugins and separate windows needed, and also in terms of analyzing one signal with relation to activity in another (f.ex. for cross-bleed reduction and/or masking features etc).

Stability

Josh asks about the stability issues reported. any special feature extractors, or other elements that have been identified that triggers instabilities. Oeyvind/Victor discuss a bit about the Python interface, as this is one issue that frequently come up in relation to compatibility and stability. There are things to try, but perhaps the most promising route is to try to get rid of the Python interface. Josh also asks about the preferred DAW used in the project, as this obviously influence stability. Oeyvind has good experience with Reaper, and this coincides with Josh’s experience at QMUL. In terms of stability and flexibility of routing (multichannel), Reaper is the best choice. Crossadaptive work directly in Ableton Live can be done, but always involve a hack. Other alternatives (Plogue Bidule, Bitwig…) are also discussed briefly. Victor suggests selecting a reference set of tools, which we document well in terms of how to use them in our project. Reaper has not been stable for Bernt and Trond, but this might be related to  setting of specific options (running plugins in separate/dedicated processes, and other performance options). In any case, the two DAWs of immediate practical interest is Reaper (in general) and Live (for some performers). An alternative to using a DAW to host the Analyzer might also be to create a standalone application, as a “server”, sending control signals to any host. There are good reasons for keeping it within the DAW, both as session management (saving setups)  and also for preprocessing of input signals (filtering, dynamics, routing).

Simplify

Some of the stability issues can be remedied by simplifying the analyzer, getting rid of unused features, and also getting rid of the Python interface. Simplification will also enable use for less trained users, as it enable self-education and ability to just start using it and experiment. Modularity might also enhance such self-education, but a take on “modularity” might simply hiding irrelevant aspects of the GUI.
In terms of feature selection the filtering of GUI display (showing only a subset) is valuable. We see also that the number of actively used parameters is generally relatively low, our “polyphonic attention” for following independent modulations generally is limited to 3-4 dimensions.
It seems clear that we have some overshoot in terms of flexibility and number of parameters in the current version of our tools.

Performative

Marije also suggests we should investigate further what happens on repeated use . When the same musicians use the same setup several times over a period of time, working more intensively, just play, see what combinations wear out and what stays interesting. This might guide us in general selection of valuable features. Within a short time span (of one performence), we also touched briefly on the issue of using static mappings as opposed to changing the mapping on the fly. Giving the system operator a more expressive role, might also solve situations where a particular mapping wears our or becomes inhibiting over time. So far we have created very repeatable situations, to investigate in detail how each component works. Using a mapping that varies over time can enable more interesting musical forms, but will also in general make the situation more complex. Remembering how performers in general can respond positively to a certain “richness” of the interface (tolerating and even being inspired by noisy analysis), perhaps varying the mapping over time also can shift the attention more on to the sounding result and playing by ear holistically, than intellectually dissecting how each component contributes.
Concluding remarks also suggests that we still need to play more with it , to become more proficient, having more control, explore and getting used to (and getting tired of) how it works.

Live convolution with Kjell Nordeson

Session at UCSD March 14.

Kjell Nordeson: Drums
Øyvind Brandtsegg: Vocals, Convolver.

Contact mikes

In this session, we explore the use of convolution with contact mikes on the drums to reduce feedback and cross-bleed. There is still some bleed from drums into the vocal mike, and there is some feedback potential caused by the (miked) drumheads resonating with the sound coming from the speaker. We have earlier used professional contact mikes, but found that our particular type did have a particularly low output, so this time we tried simple and cheap piezo elements from Radio shack, directly connected to high impedance inputs on the RME soundcard. This seems to give very high sensitivity and a fair signal to noise ratio. The frequency response is quite narrow and “characteristic” to put it mildly, but for our purposes, it can work quite well. Also, the high frequency loss associated with convolution is less of an issue when the microphones have such an abundance of high frequencies (and little or no low end).

IR update triggering

We have added the option of using a (midi) pedal to trigger IR recording. This allows a more deliberate performative control of the IR update. This was  first used by Kjell, while Øyvind was playing through Kjell’s IR. Later switched roles. Kjell notes that the progression from IR to IR works well, and that we definitely have some interesting interaction potential here. The merging of the sound from the two instruments creates a “tail” of what has been played, and that we continue to respond to that for a while.
When Kjell recorded the IR, he thought it was an extra distraction to have to also need to focus on what to record , and to operate the pedal accordingly. The mental distraction probably is not so much in the actual operation of the pedal, but in the reflection over what would make a good sound to record . It is not yet easy to foresee (hear) what comes out of the convolution process, so understanding how a particular input will work as an IR is a sort of remote and second-degree guesswork. This is of course also further complicated by not knowing what the other performer will play through the recorded IR. This will obviously become better with more experience using the techniques.
When we switched roles (vocal recording the IR), the acoustic/technical situation became a bit more difficult. The contact mikes, would pick up enough sound from the speakers (also through freely resonating cymbals resting on the drums, and via non-damped drum heads) to create feedback problems. This also creates extra repetitions of the temporal pattern of the IR due to the feedback potential. It was harder to get the sound completely dry and distinct, so the available timbral dynamic was more in the range from “mushy” to “more mushy” (…). Still, Kjell felt this was “more like playing together with another musician”. The feeling of playing through the IR is indeed the more performatively responsive situation, overpowered by the reduction in clarity that was caused by the technical/acustical difficulties. Similarly, Øyvind thought it was harder because the vocals only manifest themself as the ever changing IR, and the switching if the IR does not necessarily come across as a real/quick/responsive musical interaction. Also, delivering some material for the IR makes the quality of the material and the exact excerpt much more important. It is like giving away some part of what you’ve played, and it must be capable of being transformed out of your own control, so the material might become more transparent to it’s weaknesses. One can’t hide any flaws by stringing the material together in a well-flowing manner, rather the stringing-together is activated by the other musician. Easily, I can recognize this as the situation any musician being live sampled or live processed must feel, so it is a “payback time” revelation for me, having been in the role of processing others for many years.

Automatic IR update

We also tried automatic/periodic IR updates , as that would take the distraction of selecting IR material away, and we could more easily just focus on performing. The automatic updates shows their own set of weaknesses when compared with the manually triggered ones. The automatic update procedure essentually creates a random latency for the temporal patterns created by the convolution. This is because the periodic update is not in any way synchronized to the playing, and the performers do not have a feedback (visually or auditively) on the update pulse. This means the the IR update might happen offbeat or in the middle of a phrase. Kjell suggested further randomizing it as one solution. To this, Øyvind responds that it is already essentially random since the segmentation of input and the implied pulse of the material is unrelated, so it will shift in an unpredictable and always changing manner. Then again, following up on Kjells suggestion and randomizing it further could create a whole other, more statistical approach .  Kjell also remarks that this way of playing it feels more like “an effect”, something added, that does not respond as interactively . It just creates a (echo pattern) tail out of whatever is currently played. He suggested updating the IR at a much lower rate, perhaps once every 10 seconds. We tried a take with this setting too.

Switching who has the trigger pedal

Then, since the automatic updates seems not to work too well, and the mental distracion of selecting IR material seems unwanted, we figured, maybe the musician playing through the IR should be the one triggering the IR recording . This is similar (but exactly opposite) to the previous attempts at manual IR record triggering. Here, the musician playing through the IR is the one deciding the time of IR recording, and as such has some influence over the IR content. Still he can not decide what the other musician is playing at the time of recording , but this type of role distribution could create yet another kind of dynamic in the interplay. Sadly the session was interrupted by practical matters at this point, so the work must continue on a later occation.

Audio


Take1: Percussion IR, vocal playing through the IR. Recording/update of IR done by manual trigger pedal controlled by the percussionist. Thus it is possible to emphasize precise temporal patterns. The recording is done only with contact mikes on the drums, so there is some “disconnectedness” to the acoustic sound.


Take2: Vocal IR, percussion playing through the IR. Recording/update of the IR done by manual trigger pedal controlled by the singer. As in take 1, the drums sound going into the convolver  is only taken from the piezo pickups. Still, there is a better connectedness to the acoustic drum sound, due to an additional room mike being used (dry).


Take3: Percussion IR, automatic/periodic IR update. IR length is 3 seconds, IR update rate is 0.6 Hz.


Take4: Percussion IR, automatic/periodic IR update. IR length is 2.5 seconds, IR update rate is 0.2 Hz.

Other reflections

IR replacement is often experienced as a sudden musical change . There is no artifacts caused by the technical operation of updating the IR, but the musical result is more often a total change of “room characteristic”. Maybe we should try finding methods of slowly crossfading when updating the IR, keeping some aspects of the old one in a transitory phase. There is also a lot to be gained performatively, by the musician updating the IR having these transitions in mind. Choosing what to play and what to record is an effective way of controlling if the transitions should be sudden or slow.

Session with classical percussion students at NTNU, February 20, 2017

Introduction:

This session was a first attempt in trying out cross-adaptive processing with pre-composed material. Two percussionists, Even Hembre and Arne Kristian Sundby, students at the classical section, were invited to perform a composition written for two tambourines. The musicians had already performed this piece earlier in rehearsals and concerts. As a preparation for the session the musicians were asked to do a sound recording of the composition in order to prepare analysis methods and choice of effects before the session. A performance of the piece in its original form can be seen in this video – “Conversation for two tambourines” by Bobby Lopez performed by Even Hembre and Arne Kristian Sundby (recorded by Even Hembre).

Preparation:

Since both performers had limited experience with live electronics in general we decided to introduce the cross adaptive system gradually during the session. The session started with headphone listening, followed by introducing different sound effects while giving visual feedback to the musicians, and then performing with adaptive processing before finally introducing cross-adaptive processing. As a starting point, we used analysis methods which had already proved effective and intuitive in earlier sessions (RMS, transient density and rhythmical consonance). These methods also made it easier to communicate and discuss the technical process with the musicians during the session. The system was set up to control time based effects such as delays and reverbs, but also typical insert effects like filters and overdrive. The effect control contained both dynamical changes of different effect parameters, but also sample/hold function through the MIDIator. We had also brought a foot pedal so the performers could change the effects on the different parts of the composition during the performance.

Session:

After we had prepared and set up the system we discovered severe latency on the outputs of the system. Input signals seemed to function properly, but what was causing the latency of the output was not discovered. To solve the problem, we made a fresh set-up using the same mentioned analysing methods and effects, and after checking that the latency was gone, the session proceeded. We started with a performance of the composition without any effects, but with the performers using headphones to get familiar with the situation. The direct sound of each tambourine was panned hard left/right in the monitoring system to easier identify the two performers. After an initial discussion it was decided that both tambourines should be located in the same room since the visual communication between the performers was important in this particular piece. The microphones were separated with an acoustic barrier/screen and microphones set to cardio characteristic in order to avoid as much bleeding between the two as possible. During the performance the MIDIator was adjusted to the incoming signals. It became clear that there were some issues with bleeding already at this stage affecting the analyser, but we nevertheless retained the set-up to maintain the focus on the performance aspect. The composition had large variations in dynamics, and also in movement of the instruments. This was seen as a challenge considering the microphones’ static placements and the consequently large differences in input signal. Because of the movement, just small distance variations between instrument and microphone would have great impact in how the analysis methods read the signals. During the set-up, the visual feedback from the screen to the performers was a very welcome contribution regarding the understanding of the set-up. While setting up the MIDIator to control the effects we tried playing through the composition again trying out different effects. Adding effects made a big impact to the performance. It became clear that the performers tried to “block out” the effects while playing in order to not loose track of how the piece was composed. In this case the effects almost created a filter between the performers and the composition resulting in a gap between what they expected and what they got. This could of course be a consequence of the effects that was chosen, but the situation demanded another angle to narrow everything down in order to create a better understanding and connection between the performance and the technology. Since the composition consisted of different parts we made a selection of one of the quieter parts where the musicians could see how their playing affected their analysers, and how this further could be mapped to different effects using the MIDIator. There was still a large amount of overlapping between the instruments into the analyser because of bleeding, so we needed to take a break and rearrange the physical set-up in the room to further clarify the connection between musical input, analyser, MIDIator and effects. Avoiding the microphone bleeding helped both the system and the musicians to clarify how the input reacted to the different effects. Since the performers were interested in how this changed the sound of their instruments we agreed to abandon the composition, and instead testing out different set-ups, both adaptive and crossadaptive.

Sound examples:

1. Trying different effects on tambourine, processing musician controlling all parameters. Tambourine 1 (Even) is convolved with a recording of water and a cymbal. Tambourine 2 (Arne Kristian) is processed with delay, convolved with a recording of small metal parts and a pitch delay.

2. Tambourine 1 (Even) is analysed using transient density. The transient density is controlling a delay plug in on tambourine 2 (Arne Kristian)

3. Tambourine 2 (Arne Kristian) is analysed by transient density controlling a send from tambourine 1 convolved with cymbal. The higher transient density the less send.

4. Keeping the mapping settings from example 2 and 3 but adding rhythmical consonance analyses on Tambourine 2 to control another send level from tambourine 1 convolving it with recording of water. The higher consonance the more send. The transient density analysis on tambourine 1 is in addition mapped to control a send from tambourine 2 convolving it with metal parts. The higher density, the more send.

Observations:

Even though we worked with a composed piece it would be a good idea to have a “rehearsal” with the performers beforehand focusing on different directions through processing. This could open up for thoughts around how to do a new and meaningful interpretation of the same composition with the new elements.

It was a good idea to record the piece beforehand in order to construct the processing system, but this recording did not have any separation between the instruments either. This resulted in preparing and constructing a system that in theory were unable to be cross adaptive since it both analysed and processed the sum of both instruments leaving much less control to the individual musicians. This aspect, also concerning bleeding between microphones in more controlled environments, challenges a concept of fully controlling a cross adaptive performance. This challenge will probably be further magnified in a concert situation preforming through speakers. The musicians also noted that the separation between microphones was crucial for the understanding of the process, and the possibility to get a feeling of control.

In retrospect, the time-based effects prepared for this session could also be changed since several of them often worked against the intention of the composition, especially the most rhythmical parts. Even noted that: “Sometimes it’s like trying to speak with headphones that play your voice right after you have said the word, and that unable you to continue”.

This particular piece could probably benefit from more subtle changes from the processing. The sum of this made the interaction aspect between the performers and the technology more reduced. This became clearer when we abandoned the composition and concentrated on interaction in a more “free” setting. One way of going further into this particular composition could be to take a mixed music approach, and “recompose” and interpret it again with the processing element as a more included part of the composition process.

In the following and final part of the session, the musicians were allowed to freely improvise while being connected to the processing system. This was experienced as much more fruitful by both performers. The analysis algorithms focusing on rhythmical aspects, namely transient density and rhythmical consonance, were both experienced as meaningful and connected to the performers’ playing. These control parameters were mapped to effects like convolution and delay (cf. explanation of sound examples 1-4). The performers focused on issues of control, the differences between “normal” and inverse mapping, headphones monitoring and microphone bleeding when discussing their experiences of the session (see the video digest below for some highlights).

Video digest from session February 20, 2017

Seminar: Mixing and timbral character

Online conversation with Gary Bromham (London), Bernt Isak Wærstad (Oslo), Øyvind Brandtsegg (San Diego), Trond Engum and Andreas Bergsland (Trondheim). Gyrid N. Kaldestad, Oslo, was also invited but unable to participate.

The meeting revolves around the issues “mixing and timbral character” as related to the crossadaptive project. As there are many aspects of the project that touches upon these issues, we have kept the agenda quite open as of yet, but asking each participant to bring one problem/question/issue.

Mixing, masking

In Oslo they worked with the analysis parameters spectral crest and flux, aiming to use these to create a spectral “ducking” effect, where the actions of one instrument could selectively affect separate frequency bands of the other instrument. Gary is also interested in these kinds of techniques for mixing, to work with masking (allowing and/or avoiding masking). One could think if it as a multiband sidechaining with dynamic bands, like a de-esser, but adaptive to whichever frequency band currently needs modification. These techniques are related both to previous work on adaptive mixing (for example at QMUL) and also partially solved by recent commecial plugins, like Izotope Neutron .
However interesting these techniques are, the main focus of our current project is more on the performative application of adaptive and crossadaptive effects. That said, it could be fruitful using these techniques, not to solve old problems, but to find new working methods in the studio as well. In the scope of the project, this kind of creative studio work can be aimed at familiarizing ourselves with the crossadaptive methods in a controlled and repeatable setting. Bernt also brought up the issue of recording the analysis signals, using them perhaps as source material for creative automation, editing the recorded automation as one might see fit. This could be an effective way of familiarization with the analyzer output as well, as it invites taking a closer look at the details of the output of the different analysis methods. Recording the automation data is straightforward in any DAW, since the analyzer output comes into the DAW as external MIDI or OSC data. The project does not need to develop any custom tools to allow recording and editing of these signals, but it might be a very useful path of exploration in terms of working methods. I’d say yes please, go for it.

Working with composed material, post production

Trond had recently done a crossadaptive session with classical musicians, playing composed material. It seems that this, even though done “live” has much in common with applying crossadaptive techniques in post production or in mixing. This is because the interactive element is much less apparent. The composition is a set piece, so any changes to the instrumental timbre will not change what is played , but rather can influence the nuances of interpretation. Thus, it is much more a one-way process instead of a dialectic between material and performance. Experts on interpretation of composed music will perhaps cringe at this description, saying there is indeed a dialogue between interpretation and composition. While this is true, the degree to which the performed events can be changed is lesser within a set composition. In recent sessions, Trond felt that the adaptive effects would exist in a paralell world, outside of the composition’s aesthetic, something unrelated added on top . The same can be said about using adaptive and crossadaptive techniques in a mixing stage of a production, where all tracks are previously recorded and thus in a sense can be regarded as a set (non-changeable) source. With regards to applying analysis and modulation to recorded material, one could also mention that the Oslo sessions used recordings of the (instruments in the session) to explore the analysis dimensions. This was done as an initial exploratory phase of the session. The aim was finding features that already exist in the performer’s output, rather than imposing new dimensions of expression that the performer will need to adapt to.

On repeatability and pushing the system

The analysis-modulator response to an acoustic input is not always explicitly controllable. This is due to the nature of some of the analysis methods, technical weaknesses that introduce “flicker” or noise in the analyzer output. Even though these deviations are not inherently random, they are complex and sometimes chaotic. In spite of these technical weaknesses, we notice that our performers often will thrive. Musicians will often “go with the flow” and create on the spot, the interplay being energized by small surprises and tensions, both in the material and in the interactions. This will sometimes allow the use of analysis dimensions/methods that have spurious noise/flicker, still resulting in a consistent and coherent musical output, due to the performer’s experience in responding to a rich environment of sometimes contradicting signals. This touches one of the core aspects of our project, intervention into the traditional modes of interplay and musical communication . It also touches upon the transparency of the technology, how much should the performer be aware of the details of the signal chain? Sometimes rationalization makes us play safe. A fruitful scenario would be aiming for analysis-modulator mappings that create tension, something that intentionally disturbs and refreshes. The current status of our research leaves us with a seemingly unlimited amount of combinations and mappings, a rich field of possibilities, yet to be charted. The options are still so many that any attempt at conclusions about how it works or how to use it seems futile. Exploration in many directions is needed. This is not aimless exploration, but rather searching without knowing what can be found.

Listening, observing

Andreas mentions is is hard to pinpoint single issues in this rich field. As observer it can be hard to decode what is happening in the live setting. During sessions, it is sometimes a complex task following the exact details of the analysis and modulation. Then, when listening to the recorded tracks again later, it is easier to appreciate the musicality of the output. Perhaps not all details of the signal chain are cleanly defined and stringent in all aspects, but the resulting human interaction creates a lively musical output. As with other kinds of music making, it is easy to get caught up in detail at time of creation. Trying to listen more in a holistic manner, taking in the combined result, is a skill not to be forgotten also in our explorations.

Adaptive vs cross-adaptive

One way of working towards a better understanding of the signal interactions involved in our analyzer-modulator system is to do adaptive modulation rather than cross-adaptive. This brings a much more immediate mode of control to the performer, exploring how the extracted features can be utilized to change his or her own sound. It seems several of us have been eager to explore these techniques, but putting it off since it did not align with the primary stated goals of crossadaptivity and interaction. Now, looking at the complexity of the full crossadaptive situation, it is fair to say that exploration of adaptive techniques can serve as a very valid manner of getting in touch with the musical potential of feature-based modulation of any signal. In it’s own right, it can also be a powerful method of sonic control for a single performer, as an alternative to a large array of physical controllers (pedals, faders, switches). As mentioned earlier in this session, working with composed material or set mixes can be a challenge to the crossadaptive methods. Exploring adaptive techniques might be more fruitful in those settings. Working with adaptive effects also brings the attention to other possibilities of control for a single musician over his or her own sound. Some of the recent explorations of convolution with Jordan Morton shows the use of voice controlled crossadaptivity as applied to a musician’s own sound. In this case, the dual instrument of voice and bass operated by a single performer allows similar interactions between instruments, but bypassing the interaction between different people, thus simplifying the equation somewhat. This also brings our attention to using voice as a modulator for effects for instrumentalists not using voice as part of their primary musical output. Although this has been explored by several others (e.g. Jordi Janner , Stefano Fasciani , and also the recent Madrona Labs “Virta” synth ) it is a valid and interesting aspect, integral to our project.