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Acousmatic study by Christoph Zimmer

This article is about the three iterations of an acousmatic study by Christoph Zimmer, which were carried out as part of the seminar “Symbolic Sound Processing and Analysis/Synthesis” with Prof. Dr. Marlon Schumacher at the HFM Karlsruhe. It covers the basic concept, ideas, subsequent iterations and the technical implementation with OpenMusic.

Responsible persons: Christoph Zimmer, Master student Music Informatics at the HFM Karlsruhe

 

Basic idea and concept:

I usually work a lot with hardware for music, especially in the field of DIY. This often coincides with the organization and optimization of the workflow associated with this hardware. When we students were given the task of producing an acousmatic study in the form of musique concrète, I was initially disoriented. Up to that point, I had only dealt a little with “experimental” music genres. To be honest, I wasn’t even aware of the existence of musique concrète up to this point. So with this task I was thrown out of my usual workflow, sound synthesis with hardware, and therefore also out of my comfort zone. Now I had to use field recordings as samples.
 
My DIY attitude intuitively led me to the decision to record the samples myself. I wanted to focus on a variation of samples. However, I was still dismissive of the idea of completely cutting myself off from my previous work. I wanted to bring a “meta-connection” to my hardware-focused work into the piece. Based on this idea, the piece “chris builds a trolley for his hardware” was created
 

The finished trolley for hardware. More pictures at: https://www.reddit.com/r/synthesizers/comments/ryyw8e/i_finally_made_a_proper_stand_for_my_synth_rack/

First iteration

The piece should therefore consist of samples that were not randomly produced or downloaded from the internet, but were created as a “by-product” of work that I actually carried out myself, in this case the construction of a trolley for music hardware. Over the course of two weeks, I used my smartphone to record the sounds that emerged as I went through the various work steps. As I made use of different materials and processing methods in these work steps, not only did a wide variation of sound textures emerge, but the macroscopic structure of the piece also formed by itself. It composed itself, so to speak. The desired meta-connection was thus created. Once the trolley was complete, it was time to start producing the piece.
 
The raw audio files of the recordings are each several minutes long. To simplify handling in OpenMusic, the individual sound elements were exported as .wav files. The DAW REAPER was used for this. The result was about 350 individual samples. These are available under the following link:
 
https://drive.google.com/file/d/1hRk4OZvNEJLkpo_bzSZxP1lwO0YlcpLy/view
 
Here are a few examples of the sound elements used:
 

 

With the samples prepared, the work in OpenMusic could now begin.
As is usual for musique concrète, the samples were to be processed with various effects to support the musical context. However, it was also important to me that these effects should not dominate in such a way that the sounds become unrecognizable and the context is lost. That’s why I had the idea of programming a workspace for the arrangement within an OpenMusic patch to make the samples dynamically editable. The “Maquette” object turned out to be ideal for this. Basically, this makes it possible to place other objects within an x-axis (time) and y-axis (parameterizable). These objects can then access their own properties in the context of the maquette. I then used these functions to create four different “Template Temporal Boxes” which use the parameterization of the maquette in different ways to apply effects to the respective samples. Using multiple templates further reduces complexity while maintaining a variation of modulation possibilities:
 
tempboxa
  • Position y –> Reverbance
  • Size y –> Playback speed
  • Random –> panning

OM Patch of the tempboxa

 
tempboxb
  • Position y –> Delay time
  • Size y –> Playback speed
  • Random –> panning

OM Patch of the tempboxb

 
 
tempboxc
  • Position y –> Tremolo speed
  • Size y –> Playback speed
  • Random –> panning

OM Patch of the tempboxc

 
 
tempboxd
  • Position y –> Lowpass cutoff frequency
  • Size y –> Playback speed
  • Random –> panning

OM Patch of the tempboxd

 

With the creation of these boxes, the composition of the piece could begin.
As already mentioned, the macroscopic structure of the construction process was to be retained. In practice, certain samples of the sections (research, sketching, steel processing, welding, steel drilling, 3d printing, wood drilling, wood sanding, painting and assembly) were selected in order to process them with the parameterized tempboxes into interesting sounding combinations, which should describe the current work step.
 
 

Detail of the maquette with arrangement

 

The result of the first iteration:

 

Second iteration

 
My goal for the second iteration was to place accents on samples that represent anchor points of the piece. More precisely, the panning used in the first iteration was to be reworked by adding a provisional Haas effect (delay between the left and right channels) to the existing logic. For this purpose, the result of the previous panning is duplicated inversely and then extended with a delay (up to 8 ms) and level adjustment, which are dynamically related to the strength of the panning. Finally, both sounds are merged and output from the tempbox.

OM Patch of the extended panning

 

The result of the first iteration:

 

Third iteration

For the third and final iteration, the task was to make the piece available for an arbitrarily selectable setup of 8 channels. The structure was not to be changed. This gave me the opportunity to work on the panning again. Instead of setting the limit of the panning randomizer to 8 channels, I came up with the idea of raising the macroscopic structure even further. I chose the following speaker setup for this:
 

Setup of the speakers (with numbering of the channels)

 
With this setup, it is possible to distribute the panning to two opposite speakers, depending on the sections of the piece. During the course of the piece, the sound should then move around the listener as a slow rotational movement.
 

Part 1 of macroscopic panning

 
 
 

Part 2 of macroscopic panning

 
 

Part 3 of macroscopic panning

 
This principle applies in parallel to the accentuation of some samples from the second iteration: while the other samples (depending on the section) are distributed to different pairs of speakers, the anchor elements remain on channels 1 and 2.
 
The final version is also available in 2-channel format:

 

Fourth iteration

In this iteration, the task was to spatialize the piece using the tools we learned in the course “Visual Programming of Space/Sound Synthesis” (VPRS) with Prof. Dr. Marlon Schumacher and Brandon L. Snyder
 
“chris builds a trolley for his hardware” was already so far developed at this point that I submitted it to Metamorphoses 2022 (a competition for acousmatic pieces). For this it was necessary to mix the piece on a 16 channel setup. Due to the imminent deadline, I had very little time to adapt the piece to the requirements. Therefore, the channels were simply doubled in REAPER and LFO panning was added to the respective pairs. Unfortunately, the piece was not accepted afterwards because the length of the piece did not meet the requirements. Since the spatialization also left a lot to be desired, I took the opportunity to use the newly learned tools to improve it.
 
I decided to discard the Metamorphoses 16-channel spatialization and return to the state of the third iteration. My goal was a spatialization that not only deals with the macroscopic structure (such as the steel processing, 3D printing…), but also with the microscopic structure, i.e. to make individual sounds more dynamic. The audio exported from OM (8 channel) served as the source material, which was then to be processed using the Ambisonics (IEM) VSTs.
 
The Ambisonics template for REAPER was used as a workspace template, as it already provided a setup for the audio busses to finally render a 5th order Ambisonics file and a binaural stereo downmix. In the first step, the 8-channel audio file was routed so that it could be processed separately. To do this, channels 1-2, 3-4, 5-6 and 7-8 were sent to new tracks and the master send was deactivated. These tracks were then defined as multi-channel tracks with 36 channels and the stereo encoder (IEM) was inserted into the effect chain. The parameters for the spatialization (azimuth, elevation, roll and width) were then added as envelopes to the REAPER timeline to enable their dynamic processing. Finally, all tracks can be merged into the Ambisonics bus. The binaural downmix was used as a monitoring output.
 

A simplified representation of the routing in REAPER

 
In practice, points were inserted into the envelope tracks by hand, between which linear interpolation was then used to create dynamic changes in the parameters. I proceeded intuitively and listened to individual sections to get a basic idea of what kind of spatialization would emphasize this section. Then I looked at the individual sounds and their origins and tried to describe them with the help of the parameters. Examples of this are: an accelerating rotary movement when drilling, a jumping back and forth when the digital input of the 3D printer beeps or a complete mess when crumpling paper. I was already familiar with this type of workflow, not only when using DSP VSTs in the DAW, but also when programming DMX lights via the envelope.
 
When editing, I found the visual feedback of the EnergyVisualizer (IEM) not only very helpful to keep an overview. I therefore decided to record it and add it to the binaural downmix:
 
 
All uncompromised files can be found under the following link: https://drive.google.com/drive/folders/1bxw-iZEQTNnO92RTCmW_l5qRFjeuVxA9?usp=sharing
 

ByLukas Körfer

Speaking Objects

Abstract

In this project, an audio-only augmented reality sound installation was created as part of the course „Studienprojekte Musikprogrammierung“ (“Study Projects Music Programming”) at the Karlsruhe University of Music. It is important for the following text to distinguish the terminology from virtual reality (VR for short), in which the user is completely immersed in the virtual world. Augmented reality (AR for short) is the extension of reality through the technical addition of information.

 

Motivation

On the one hand, this sound installation should meet a certain artistic standard, on the other hand, my personal goal was to bring AR and especially auditory AR closer to the participants and to get them excited about this new technology. Unfortunately, augmented reality is very often only understood as the visual representation of information, as is the case with navigation systems or smartphone applications, for example. However, in my opinion, it is important to sensitize people more and more to the auditory extension of reality. I am convinced that this technology also has enormous potential and that there is a lot of catching up to do in terms of public awareness compared to visual augmented reality. There are already numerous areas of application in which the benefits of auditory AR have been demonstrated. These range from areas in which many applications of visual AR can already be found, such as education, increasing productivity or purely for entertainment purposes, to specialist areas such as medicine. Ten years ago, for example, there were already attempts to use auditory AR to enhance the sense of hearing for people with visual impairments. By sonifying real objects, it was possible to create a purely auditory orientation aid.

 

Methodology

In this project, participants should be able to move freely in a room in which objects are positioned and although these do not produce sounds in reality, the participants should be able to perceive sounds through headphones. In this sense, it is an extension of reality (“augmented reality”), as information is added to reality in auditory form using technical means. Essentially, the areas for implementation extend on the one hand to the positioning of the person (motion capture) and binauralization and on the other hand in the artistic sense to the design of the sound scene by positioning and synthesizing the sounds.

Figure 1

The motion capture in this project is realized with the Polhemus G4 system. The direction and position of a micro-sensor, which is attached to a pair of glasses worn by the participant, is determined by a magnetic field generated by two transmitters. A hub, which is connected to the micro-sensor via a cable, sends the motion capture data to a USB dongle connected to a laptop. This data is sent to another laptop, on which the binauralization takes place and which is ultimately connected to the wireless headphones.

Figure 2 shows two of the six objects in one variant each (angles of 45° and 90°). The next illustration (Fig. 3) shows the over-glasses (protective glasses that can also be worn over glasses) that are used in the sound installation. These goggles have a wide nose bridge to which the micro-sensor is attached with a micro-mount from Polhemus.

Figure 2

 

Figure 3

As previously explained, various decisions have to be made before the artistic aspect of the sound installation can be realized. This involves the positioning of the objects / sound sources and the sounds themselves.

Figure 4

 

Figure 5

Figure 4 shows a sketched top view of the complete structure. The six blue-colored circles mark the positions of the objects in the room and, of course, the sound sources of the scene in Binauralix, which can be seen in Figure 5. The direction and angle of the sources can be taken from the colorless areas (in Fig. 4), at either 45° or 90° angles, around the sound sources.

The completely wireless position detection and data transmission enables the participants to immerse themselves fully in this experience of the interactive reality-expanding sound world. The sound synthesis was carried out using the SuperCollider software. The sounds were mainly created through various tapping and clicking noises recorded by the SoundIn object, and finally changes and alienation of the sounds through amplitude and frequency modulation and various filters. By routing the sounds to a total of 6 output channels and “s.record(numChannels:6)”, I was able to create a two-minute multi-channel audio file in SuperCollider. When playing the file in Binauralix, the first channel is automatically mapped to source one, the second channel to source 2 and so on.

 

Technical implementation

The technical challenge for the implementation of the project initially consisted of receiving and reformatting the data from the sensor so that it could be used in Binauralix. The initial problem was that Binauralix is only available for MacOS and the software for the Polhemus G4 system is only available for Windows and Linux. As I had a MacBook and a laptop with Ubuntu Linux as my operating system at the time, I installed the Polhemus software for Linux.

After building and installing the Polhemus G4 software on Linux, the five applications “G4DevCfg”, “CreateSrcCfg”, “g4term”, “g4display” and “g4export” were available. For my project, all devices used must first be connected and configured with “G4DevCfg”. The terminal application “g4export” can be used to transmit the sensor data via UDP by specifying the previously created source configuration file, the local IP address of the receiver device and a port. The source configuration file is a file in which the position and orientation of the transmitter are defined by a “virtual frame of reference” and settings can be made for the entry hemisphere into the magnetic field, floor compensation and source calibration file. To run the application, the transmitters and the hub must be switched on at this point, the USB dongle must be connected to the laptop and the sensor to the hub, and the hub must be connected to the USB dongle. If the MacBook is now in the same network as the Linux laptop, the data can be received by specifying the previously used port. This is done with my sound installation in a self-created MaxMSP patch.

Figure 6

In this application, the appropriate port must first be selected on the left-hand side. As soon as the connection is established and the messages arrive, you can view them in raw form under the selection field. The six values that can be seen at the top in the middle of the application are the values for position and orientation that have been separated from the raw message. Final settings for the correct calibration can now be made in the action field below. There is also the option to mirror the axes individually or to change the Yaw value if unexpected problems should arise when setting up the sound installation. Once the values have been formatted into messages that can be used by Binauralix (visible at the bottom right of the application), they are sent to Binauralix.

The following videos provide a view of the scene in Binauralix and an auditory impression as the listener — driven by the sensor data — moves through the scene.

 

 

Past performances of the sound installation

The sound installation as a contribution to the EFFEKTE lecture series of the Wissenschaftsbüro-Karlsruhe

 

 

test
The sound installation as the subject of a workshop for the Kulturakademie at the HfM-Karlsruhe

 

ByLorenz Lehmann

“OM-LEAD” library

Abstract:

The library “OM-LEAD” is a library for rule-based, computer-generated real-time composition. The considerations and approaches in Joseph Branciforte’s text “FROM THE MACHINE: REALTIME ALGORITHMIC APPROACHES TO HARMONY AND ORCHESTRATION” are the starting point for the development.

The library currently comprises two functions that are written using both CommonLisp and existing functions from the OM package.

In addition, the composition is currently limited in the scope of the parameters to be controlled to the harmonics and the voice leading.

In the future, I would also like to write a function that allows composition on a temporal level with the parameters metrics and cue spacing.

Development: Lorenz Lehmann

Supervision and advice: Prof. Dr. Marlon Schumacher

My sincere thanks for their kind support go to Joseph Branciforte and

Prof. Dr. Marlon Schumacher.

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