Tag Archive OpenMusic

ByFlorian Simon

PixelWaltz: Sonification of images in OpenMusic

Abstract: The OpenMusic program PixelWaltz can be used to convert images into symbolic representations of music (pitches and onset times). Options for image manipulation are available with which the result can be additionally influenced.

Responsible persons: Florian Simon

Mapping: Pitch

The pixels of the image are scrolled through line by line and the respective red, green and blue values (between 0 and 1) are mapped to a desired pitch range. This means that three pitch values in midicent are always obtained from one pixel. As two adjacent pixels are similar in many cases, this mapping method often results in repeating patterns every three notes. This is the reason for the title of the project.

It is also possible to limit the number of note values output.

Mapping: Application times

A constant value can be set for the start times and note durations. A humanizer effect can also be switched on, which randomly shifts each note forwards or backwards within a specified range. Starting from the basic tempo, accelerandi and ritardandi can be created by passing lists of three numbers. These represent the start note, end note and speed of the tempo change. (20 50 -1) creates an accelerando from note 20 to note 50, in which the intervals per note become one millisecond shorter. A positive third value corresponds to a ritardando.

Dynamics

Different random ranges for “red”, “green” and “blue” notes can be defined for the volume or velocity. The values generated in this way can also be modulated sinusoidally so that, for example, the volume can rise and fall over longer periods of time. This requires the specification of a wavelength in the number of notes and the maximum deviation factor.

Accompaniment

PixelWaltz offers the option of generating an accompanying voice, which consists of individual additional tones in a desired fixed note number frequency. If this is not divisible by 3, a polymetric is often created. The pitch is determined randomly and can be between 3 and 6 semitones below the respective “accompanied” note.

Image processing

In order to create further variation, the sonification section of PixelWaltz is preceded by tools for manipulating the input image. In addition to adjusting the image size, brightness and contrast, it is also possible to shift the color values and thus recolor the image. The changes in the musical translation are immediately noticeable: More brightness leads to a higher average pitch, more contrast reduces the number of different pitch values. With a blue-dominated image, the last notes of the triplet will usually be the highest.

Sound results

The tonal results naturally differ depending on the input – but photographed material in particular often leads to the same wave-like overall structure, which winds irregularly and at a slow tempo chromatically, sometimes upwards, sometimes downwards. The accompaniment supports this effect and can form a counter-pulse to the main voice.

ByLaura Peter

Whitney Music Box with OMChroma/OMPrisma in OpenMusic

The Whitney Music Box is a sonified and/or visual representation of a series of interrelated sound elements. From a musical point of view, these elements can be related chromatically or harmonically, for example. In the visual representation, each of these elements is represented by a circle or dot (see Figure 1). These dots circle around a common center point depending on their own assigned frequency. The lower the frequency, the smaller the radius of the orbiting circle and the higher the orbital speed. Each sound element represents multiples of a fixed fundamental frequency in a harmonic series. As soon as an element has completed a revolution around the center point, the sound is triggered with the frequency it represents. Due to the mathematical relationship between the individual elements, there are moments during the performance of the Whitney Music Box in which certain elements are triggered simultaneously and phases in which the elements can be perceived consecutively. At the beginning and at the end, all elements are triggered simultaneously.

Figure 1: Whitney Music Box – visual representation

In this project, OMChroma is used to synthesize the individual sound elements (see Figure 2). The synthesis classes of OMChroma inherit from OpenMusic’s class-array object. The columns in the array describe the individual components within the synthesis. The rows represent parameters that can be assigned locally to the individual components or globally to the entire process. For the Whitney Music Box, elements are needed that implement the individual pitch gradations and the temporal offset of the individual pitch gradations. An OMChroma matrix is regarded as an event. Such an event represents a pitch and the sound repetitions within the global duration of the Whitney Music Box. The global duration is defined at the beginning and also describes the round trip time of the lowest frequency or the previously defined start frequency. Each matrix represents a frequency that is a multiple of the start frequency. The round trip time of a sound element is calculated using the formula

duration(global) / n

Where n is the index of the individual sound elements or matrices. The higher the index, the higher the frequency and the shorter the round trip time. The repetitions of the sound elements are defined by the parameter e-dels . Each component of a matrix is given a different entry delay. These entry delays are spaced at regular intervals of duration(global) / n.

Figure 2: Application of OMChroma

Without spatialization, the Whitney Music Box with OMChroma sounds like this:

Alex Player - Best audio player

Figure 3 shows how the collected matrices or sound events are spatialized with the OMPrisma library. This was based on the visual representation of the Whitney Music Box. Sound elements with a low frequency are further away from the center and sound elements with a high frequency circle closer to the center. With OMPrisma, this representation is to be implemented in spatial sound. This means that sounds with a low frequency should sound further away and sounds with a high frequency should sound closer to the listener. In the OpenMusic patch, elements with an even index were also positioned further to the front and further to the right and, similarly, elements with an odd index were positioned further to the left and back in order to distribute the sounds evenly in the room. The OMPrisma classes also offer presets for the attenuation function, air-absorption function and time-of-flight function . These were used to create an even greater sense of spatiality in addition to the positioning in the room.

Figure 3: Application of OMPrisma

In stereo, for example, the Whitney Music Box sounds like this:


Figure 4 shows how the collected OMChroma and OMPrisma matrices are merged using the chroma-prisma function. The list of all collected matrices is returned via an om-loop and rendered as a sound using the synthesize function(see Figure 5).

Figure 4: chroma-prisma

Figure 5: loop and synthesize

The OpenMusic patch and sound samples can be downloaded from the following link: https://github.com/lauraptrcodes/Whitney-music-box

ByMoritz Reiser

Markov processes for controlling harmonics in OpenMusic and Common Lisp

Abstract: A project on the use of random processes in a musical context. Basically, two different models are used. These generate chord sequences, which are then provided with a rhythm and an overlying melody.

Responsible: Moritz Reiser

 

Overview

The overall structure of the program, which corresponds to the content of the main patch, can be seen in Figure 1. At the top is the selection of the algorithm to be used for chord progression generation. This can be selected via the selection field at the top left. The two input fields of the subpatches can be used to specify the desired length and the starting chord or the key of the composition.

This is followed by a random determination of the respective tone lengths. Here you can set the tempo in BPM as well as the frequencies of the tone lengths occurring in multiples of quarter notes. The respective start times of the chords are calculated from the calculated durations using a “dx→x” function. When using the program, care must be taken here that Open Music calculates new random numbers in both strings due to the output being used twice, as a result of which the relationship between the start time and the tone duration is lost. This can be remedied by locking the subpatches for chord progression and tone length generation with “Lock Eval” after running the program once and then running it again to adjust the start times to the now saved tone durations (see information panel in the main patch). The third major step in the overall process is the generation of a melody that lies above the chord sequence. Here, a note is selected from the underlying chord and shifted up an octave. You can set whether this should always be a random chord tone or whether the tone closest to or furthest away from the preceding melody tone should be selected.

The result is then visualized at the bottom in a multi-seq object.

Figure 1: Overall structure of the composition process

 

Chord progression generation

Two algorithms are available for generating the chord sequence. The desired length of the sequence, which corresponds to the number of chords, and the starting chord or the key are transferred to them.

Harmonic chord sequence using Markov chain

The sequence of the first algorithm can be seen in Figure 2. The subpatch “Create Harmonic Chords” generates the basic set of chords that will be used in the following. This corresponds to the usual levels of counterpoint theory and, in addition to the tonic, subdominant, dominant and their parallels, contains a diminished chord on the seventh degree, a sixth ajoutée of the subdominant and a dominant seventh chord. The “Key” input adds a value corresponding to the desired key to these chords.

Figure 2: Subpatch for generating a harmonic chord sequence using a Markov chain

The “Create Transition Matrix” subpatch generates a matrix with transition probabilities for the individual chords. For each chord step, the probability with which it transitions to a certain other chord is determined. The probability values were chosen arbitrarily according to the usual processes in counterpoint theory and adjusted experimentally. For each chord it was investigated how likely it is to transition from this chord to another chord, so that the result corresponds to the conventions of counterpoint theory and allows a frequent return to the tonic level in order to focus on it. The exact transition probabilities are listed in the following table, whereby the initial sounds are listed in the left-hand column and the transitions are represented line by line.

Table 1. Transition probabilities of the harmonies of corresponding chord levels

The generation of the chord sequence finally takes place in the patch “Generate Markov Series”, which is shown in Figure 3. This initially only works with the numbering of the chord steps, which is why it is sufficient to pass it the length of the chord list. The Lisp function “Markov Synthesis” now generates a chord sequence of the desired length using the transition matrix. As it is not guaranteed that the last chord in the sequence generated in this way corresponds to the tonic, another Lisp function is used, which generates further chords until the tonic is reached. As the steps have only been numbered so far, the chords valid for the respective steps are finally selected in order to obtain the finished chord sequence.

Figure 3: Subpatch for generating a chord sequence using Markov synthesis

 
Chromatic chord progression using a tone net

In contrast to the harmonic chord progression, all 24 major and minor chords of the chromatic scale are used here (see Figure 4). The special feature of this algorithm lies in the choice of transition probabilities. These are based on a so-called tone network, which is shown in Figure 5.

Figure 4: Subpatch for generating a chord sequence based on the tone net representation

Figure 5: Tonnetz (image source:<https://jazz-library.com/articles/tonnetz/>)

Within the tone net, individual tones are applied and connected to each other. On the horizontal lines, the tones are each a fifth apart, while the diagonal lines show minor thirds (from top left to bottom right) and major thirds (from bottom left to top right). The resulting triangles each represent a triad, for example the triangle of the notes C, E and G results in the chord C major. All major and minor chords of the chromatic scale can be found. The Tonnetz representation is mostly used for analysis purposes, as a Tonnetz allows you to see directly how many tones two different triads share. One example is the analysis of classical music of the romantic and modern periods as well as film music, as the harmonic counterpoint rules used above are often neglected here in favor of chromatic and other previously unusual transitions. The distance between two chords in the tonal network can be a measure of whether the transition of one chord into the other is melodious or rather unusual. It is calculated from the number of edges that have to be crossed to get from one chord triangle to another. In other words, it corresponds to the degree of adjacency between two triangles, whereby a direct adjacency results from sharing an edge. Figure 6 shows an example of this: To get from the chord C major to the chord F minor, three edges have to be crossed, resulting in a distance of 3.

Figure 6: Example of determining the distance in the tone network using the transition from C major to F minor

As part of the project, the transition probabilities are now calculated on the basis of the distances between chords in the tone network. It is only necessary to distinguish whether the active triad is a major or minor chord, as the same distances to other chords result for all keys within these two classes. This means that every transition can be calculated from C major or C minor and then shifted to the desired key by adding a value. Starting from both variants (C major and C minor), the distances to all other triads were first recorded in the tonal network:

Distances from C major:

Intervals from C minor:

In order to obtain probabilities from the intervals, all values were first subtracted from 6 to make larger intervals less probable. The results were then used as the exponent of the number 2 in order to give greater weighting to closer chords. Overall, this results in the formula

P=2^(6-x) ; P=probability, x=distance in the grid

to calculate the transition weights. These result in the following matrix for all possible chord combinations, from which 342 probabilities result when divided by the row sum.

Within the patch, the Lisp function “Generate Tonnetz Series” first determines whether the active chord is a major or minor triad. As with the harmonic procedure, only the numbers 0-23 are used initially, this can be determined using a simple modulo-2 calculation. Depending on the result, the respective probability vector is used, a new chord is determined and finally the previous step is added. If the result is a number greater than 23, 24 is subtracted in order to always remain within the same octave.

After the previously determined length of the sequence, this section is finished. There is no return to the tonic as in the previous section, as the chromaticism means that the tonic is not as pronounced as in the harmonic chord sequence.

Determining the tone lengths

After a chord progression has been generated, random lengths are calculated for the individual triads. This is done in the “Calculate Durations” subpatch, which is shown in Figure 6. In addition to the desired BPM number, a list of note lengths is transferred as multiples of quarter notes. More probable values occur more frequently in this pool, so that a corresponding selection can be made via “nth-random”.

Figure 7: Subpatch for random determination of note durations

Melody generation

The basic melody generation process has already been described above: A tone is selected from the respective chord and transposed up an octave. This tone can be selected at random or according to the smallest or largest distance to the previous tone.

 

Sound examples

Example of a harmonic chord sequence:

 
 

Example of a tone net chord sequence:

 
 

 

ByAndres Kaufmes

Transient Processor

Transient Processor

SKAS symbolic sound processing and analysis/synthesis

Prof. Dr. Marlon Schumacher

Intermediate project by Andres Kaufmes

HfM Karlsruhe – IMWI (Institute for Music Informatics and Musicology)

Winter semester 2022/23

_____________

For this interim project, I worked on the implementation of a transient processor in OpenMusic with the help of the OM-Sox library.
A transient processor (also known as a transient designer or transient shaper) can be used to influence the attack/release behavior of the transients of an audio signal.

The first hardware device presented was the SPL TD4, introduced by SPL in 1998, which was available as a 19″ rack device and is still available today in an advanced version.

Transient Designer from SPL. (c) SPL

Transient Designers are particularly suitable for processing percussive sounds or speech. First, the transients must be isolated from the desired audio signal; this can be done using a compressor, for example. A short attack time “ducks” the transients and the signal can be subtracted from the original. The audio signal can then be processed with further effects in the course of the signal chain.

Transient processor patch. FX chain of the two signal paths (left “Transient”, right “Residual”).

At the top of the patch you can see the audio file to be processed, from which, as just described, the transients are isolated using a compressor and the resulting signal is subtracted from the original. Now two signal paths are created: The isolated transients are processed in the left-hand “chain”, the residual signal in the right-hand one. After both signal paths have been processed with audio effects, they are mixed together, whereby the mixing ratio (dry/wet) of both signal paths can be adjusted as desired. At the end of the signal processing there is a global reverb effect.

“Scope” view of the two signal paths. Sketches of the possible signal path and processing.

Sound examples:

Isolated signal:

Residual signal:

Byadmin

Spatial transformation of the piece “Ode An Die Reparatur” (“Ode To The Repair”)

Abstract: The entry describes the spatialization of the piece “Ode An Die Reparatur” (“Ode To The Repair”) (2021) and its transformation into a Higher Order Ambisonics version. A binaural mix of the finished piece makes it possible to understand the working process based on the result.

Supervisor: Prof. Dr. Marlon Schumacher

A contribution by: Jakob Schreiber

Piece

The piece “Ode An Die Reparatur” (2021) consists of four movements, each of which refers to a different aspect of a fictional machine. What was interesting about this process was to investigate the transition from machine sounds to musical sounds and to shape it over the course of the piece.

Production

The production resources for the piece were, on the one hand, a UHER tape recorder, which enables simple repitch changes and was predestined for the realization of this piece based on mechanical machines due to its functionality with motors and belts. SuperCollider was also used as a digital sound synthesis and alienation environment.

Structure

The piece consists of four movements.

First movement

The sound material of the beginning is composed of various recordings from a tape recorder, over which clearly audible, synthesized engine sounds are played alongside silence.

Second movement

The sound objects, some of which are reminiscent of birdsong, suddenly emerge from sterile silence into the foreground.

Third movement

The perforative characteristics inside a gearwheel are transformed into tonal resonances in the course of the movement.

Fourth movement

In the last movement, the engines play a monumental final hymn.

Spatialization

Based on the compositional form of the piece, the spatialization drafts adhere to the division into movements.

Working practice

The working process can be divided into different sections, similar to the OM patch. In the laboratory section, I explored different forms of spatialization in terms of their aesthetic effect and examined their conformity with the compositional form of the existing piece.

In order to determine different trajectories, or fixed positions of sound objects, the visual assessment of the respective trajectories played an important role in addition to the auditory effect.

Ultimately, the parameters of the pre-selected trajectories were supplemented with a scattering curve, finely adjusted and finally transformed into a fifth-order Higher Order Ambisonics audio file via a chain of modules.

Iteratively, the synthesized multi-channel files are added to the overall structure in REAPER and their effect is examined before they are run through the synthesis process again with an optimized set of parameters and trajectories.

More details on the individual sentences

First, you can listen to the binaural version of the spatialized piece as a whole. The approaches of the individual parts are described briefly below

 
First sentence

In the first part, the long, drawn-out clouds of sound, lying on top of each other like layers, move according to the basic tempo of the movement. The trajectories lie in a U-shape around the listening area, covering only the sides and front.

Second movement

Individual sound objects should be heard from very different positions. Almost percussive sounds from all directions of the room make the listener’s attention jump.

Third movement

The sonic material of this part is a sound synthesis based on the sound of cogwheels or gears. The focus here was on immersion in the fictitious machine. From this very sound material, resonances and other changes create horizontal tones that are strung together to form short motifs.

The spatialization concept for this part is made up of moving and partly static objects. The moving ones create an impression of spatial immersion at the beginning of the movement. At the end of the movement, two relatively static objects are added to the left and right of the stereo base, which primarily emphasize the melodic aspects of the sound and merely oscillate fleetingly on the vertical axis at their respective positions.

Fourth movement

The instrumentation of this part of the piece consists of three simulations of an electric motor, each of which follows its own voice. In order to separate the individual voices a little better, I decided to treat each of the four motors as an individual sound object. To support the monumental character of the final part, the objects only move very slowly through the fictitious space.

 

Byadmin

Extension of the acousmatic study – 3D 5th-order Ambisonics

This article is about the fourth iteration of an acousmatic study by Zeno Lösch, which was carried out as part of the seminar “Visual Programming of Space/Sound Synthesis” with Prof. Dr. Marlon Schumacher at the HFM Karlsruhe. The basic conception, ideas, iterations and the technical implementation with OpenMusic will be discussed.

Responsible persons: Zeno Lösch, Master student Music Informatics at HFM Karlsruhe, 2nd semester

 

Pixel

A Python script was used to obtain parameters for modulation.

This script makes it possible to scale any image to 10 x 10 pixels and save the respective pixel values in a text file. “99 153 187 166 189 195 189 190 186 88 203 186 198 203 210 107 204 143 192 108 164 177 206 167 189 189 74 183 191 110 211 204 110 203 186 206 32 201 193 78 189 152 209 194 47 107 199 203 195 162 194 202 192 71 71 104 60 192 87 128 205 210 147 73 90 67 81 130 188 143 206 43 124 143 137 79 112 182 26 172 208 39 71 94 72 196 188 29 186 191 209 85 122 205 198 195 199 194 195 204 ” The values in the text file are between 0 and 255. The text file is imported into Open Music and the values are scaled.

These scaled values are used as pos-env parameters.

Reaper and IEM-Plugin Suite

 

With different images and different scaling, you get different results that can be used as parameters for modulation. In Reaper, the IEM plug-in suite was used in post-production. These tools are used for Ambisonics of different orders. In this case, Ambisonics 5 order was used. One effect that was often used is the FDNReverb. This reverb unit offers the possibility of applying an Ambisonics reverb to a multi-channel file. The stereo and mono files were first encoded in 5th order Ambisonics (36 channels) and then converted into two channels using the binaural encoder. Other post-processing effects (Detune, Reverb) were programmed by myself and are available on Github. The reverb is based on a paper by James A. Moorer About this Reverberation Business from 1979 and was written in C. The algorithm of the detuner was written in C from the HTML version of Miller Puckette’s handbook “The Theory and Technique of Electronic Music”. The result of the last iteration can be heard here.

Byadmin

BAD GUY: An acousmatic study

Abstract:

Inspired by the “Infinite Bad Guy” project, and all the very different versions of how some people have fueled their imaginations on that song, I thought maybe I could also experiment with creating a very loose, instrumental cover version of Billie Eilish’s “Bad Guy”.

Supervisor: Prof. Dr. Marlon Schumacher

A study by: Kaspars Jaudzems

Winter semester 2021/22
University of Music, Karlsruhe

To the study:

Originally, I wanted to work with 2 audio files, perform an FFT analysis on the original and “replace” its sound content with content from the second file, based only on the fundamental frequency. However, after doing some tests with a few files, I came to the conclusion that this kind of technique is not as accurate as I would like it to be. So I decided to use a MIDI file as a starting point instead.

Both the first and second versions of my piece only used 4 samples. The MIDI file has 2 channels, so 2 files were randomly selected for each note of each channel. The sample was then sped up or down to match the correct pitch interval and stretched in time to match the note length.

The second version of my piece added some additional stereo effects by pre-generating 20 random pannings for each file. With randomly applied comb filters and amplitude variations, a bit more reverb and human feel was created.

Acoustic study version 1

Acousmatic study version 2

The third version was a much bigger change. Here the notes of both channels are first divided into 4 groups according to pitch. Each group covers approximately one octave in the MIDI file.

Then the first group (lowest notes) is mapped to 5 different kick samples, the second to 6 snares, the third to percussive sounds such as agogo, conga, clap and cowbell and the fourth group to cymbals and hats, using about 20 samples in total. A similar filter and effect chain is used here for stereo enhancement, with the difference that each channel is finely tuned. The 4 resulting audio files are then assigned to the 4 left audio channels, with the lower frequency channels sorted to the center and the higher frequency channels sorted to the sides. The same audio files are used for the other 4 channels, but additional delays are applied to add movement to the multi-channel experience.

Acousmatic study version 3

The 8-channel file was downmixed to 2 channels in 2 versions, one with the OM-SoX downmix function and the other with a Binauralix setup with 8 speakers.

Acousmatic study version 3 – Binauralix render

Extension of the acousmatic study – 3D 5th-order Ambisonics

The idea with this extension was to create a 36-channel creative experience of the same piece, so the starting point was version 3, which only has 8 channels.

Starting point version 3

I wanted to do something simple, but also use the 3D speaker configuration in a creative way to further emphasize the energy and movement that the piece itself had already gained. Of course, the idea of using a signal as a source for modulating 3D movement or energy came to mind. But I had no idea how…

Plugin “ambix_encoder_i8_o5 (8 -> 36 chan)”

While researching the Ambix Ambisonic Plugin (VST) Suite, I came across the plugin “ambix_encoder_i8_o5 (8 -> 36 chan)”. This seemed to fit perfectly due to the matching number of input and output channels. In Ambisonics, space/motion is translated from 2 parameters: Azimuth and Elevation. Energy, on the other hand, can be translated into many parameters, but I found that it is best expressed with the Source Width parameter because it uses the 3D speaker configuration to actually “just” increase or decrease the energy.

Knowing which parameters to modulate, I started experimenting with using different tracks as the source. To be honest, I was very happy that the plugin not only provided very interesting sound results, but also visual feedback in real time. When using both, I focused on having good visual feedback on what was going on in the audio piece as a whole.

Visual feedback – video

Channel 2 as modulation source for azimuth

This helped me to select channel 2 for Azimuth, channel 3 for Source Width and channel 4 for Elevation. If we trace these channels back to the original input midi file, we can see that channel 2 is assigned notes in the range of 110 to 220 Hz, channel 3 notes in the range of 220 to 440 Hz and channel 4 notes in the range of 440 to 20000 Hz. In my opinion, this type of separation worked very well, also because the sub-bass frequencies (e.g. kick) were not modulated and were not needed for this. This meant that the main rhythm of the piece could remain as a separate element without affecting the space or the energy modulations, and I think that somehow held the piece together.

Acousmatic study version 4 – 36 channels, 3D 5th-order Ambisonics – file was too big to upload

Acoustic study version 4 – Binaural render

Byadmin

Spectral Select: An acousmatic 3D audio study

 

 

Abstract:
Spectral Select explores the spectral content of one sample and the amplitude curve of a second sample and unites them in a new musical context. The meditative character of the output created by iteration is both contrasted and structured by louder amplitude peaks.
In a revised version, Spectral Select was spatialized in Ambisonics HOA-5 format.

Supervisor: Prof. Dr. Marlon Schumacher

A study by: Anselm Weber

Winter semester 2021/22
University of Music, Karlsruhe

 


About the study:
In which forms of expression is the connection between frequency and amplitude expressed ? Are both areas intrinsically connected and if so, what could be approaches to redesigning this order?
Such questions have occupied me for some time. That’s why the attempt to redesign them is the core topic of Spectral Select.
I was inspired by AudioSculpt from IRCAM, which we got to know in our course: “Symbolic Sound Processing and Analysis/Synthesis” together with Prof. Dr. Marlon Schumacher and Brandon L. Snyder and which we partially rebuilt.

Spectral Edit works on a similar principle, but instead of having a user work out interesting areas within a spectrum of a sample, it was decided to use a second audio sample. This additional sample (from now on referred to as “amplitude sound” in the course of this article) determines how the first sample (from now on referred to as “spectral sound”) is to be processed by OM-Sox.
To achieve this, two loops are used:
First, individual amplitude peaks are analyzed out of the amplitude sound in the first “peakloop”. This analysis is then used in the heart of the patch, the “choosefreq” loop, to select interesting sub-ranges from the spectral sample. Loud peaks filter narrower bands from higher frequency ranges and form a contrast to weaker peaks, which filter somewhat broader bands from lower frequency ranges.

peakloop – Analysis
choosefreq Loop – Audio Processing


How small the respective iteration steps are affects both the length and the resolution of the overall output. Depending on the sample material, a large number of short grains or fewer but longer subsections can be created. However, both of these parameters can be selected freely and independently of each other.

In the enclosed piece, for example, a relatively high resolution (i.e. an increased number of iteration steps) was chosen in combination with a longer duration of the cut sample. This creates a rather meditative character, whereby no two sections will be 100% identical, as there are constantly minimal changes under the peak amplitudes of the amplitude sound.
The still relatively raw result of this algorithm is the first version of my acousmatic study.

Acousmatic study version 1


The subsequent revision step was primarily aimed at working out the differences between the individual iteration steps more precisely. For this purpose, a series of effects were used, which in turn behave differently depending on the peak amplitude of the amplitude sound. To make this possible, the series of effects was integrated directly into the peak loop.

Acousmatic study version 2


In the third and final revision step, the audio was spatialized to 8 channels.
The individual channels sound into each other and change their position in a clockwise direction. This means that the basic character of the piece remains the same, but it is now also possible to follow the “working through” of the choosefreq loop spatially. To maintain this spatiality, the output was then converted to binaural stereo for the upload using Binauralix.

Acoustic study version 3 – Binaural

 

Spectral Select – Ambisonics

In the course of a further revision, Spectral Select was re-spatialized using the spatialization class “Hoa-Trajectory” from OM-Prisma and converted to the Ambisonics format.
To ensure that this step fits in well conceptually and sonically with the previous edits, the amplitude sound should also play an important role in the spatial position.
The possibilities for spatializing sounds with the help of Open Music and OM Prism are numerous. In the end, it was decided to work with Hoa-Trajectory. Here, the sound source is not bound to a fixed position in space and can be described with a trajectory that is scaled to the total duration of the audio input.

Spatialization with HOA.TRAEJECTORY

 

 

The trajectory is created depending on the amplitude analysis in the previous step.
A simple, three-dimensional circular movement, which spirals downwards, is perturbed with a more complex, two-dimensional curve. The Y-values of the more complex curve correspond to the analyzed amplitude values of the amplitude sound.
Depending on the scaling of the amplitude curve, this results in more or less pronounced deviations in the circular motion. Higher amplitude values therefore ensure more extensive movements in space.

 

 


It is interesting to note that OM-Prisma also takes Doppler effects into account. As a result, it is also audible that at higher amplitude values, more extreme distances to the listening position are covered in the same time. This step therefore has a direct influence on the timbre of the entire piece.
Depending on the scaling of the trajectory, fast movements can be strongly overemphasized, but artifacts can also occur (if the distance is too great).
To get a better impression, 2 different runs of the algorithm with different distances to the listener follow.

 

Version with extreme Doppler effects which can result in artifacts – binaural stereo

Versionwith closer distance and more moderate Dopp ler effects – Binaural Stereo

 

In contrast to the previous sound examples, the spectral sound and amplitude sound have been replaced in this example. This is a longer sound file for analyzing the amplitudes and a less distorted drone as a spectral sound.
The idea behind this project is to experiment with different sound files anyway.
Therefore, the old algorithm has been reworked to offer more flexibility with different sound files:

Revised scalable version of the old algorithm for selecting from the spectral sound

In addition, a randomized selection is now made from the spectral sound on the time axis. As a result, any shaping context should come from the magnitude of the amplitude sound and any timbre should be extracted from the spectral sound.

 

ByVeronika Reutz

Composing in 8 channels with OpenMusic

In this article I present my ideas, creative processes and technical data for the patch programmed for the class “Symbolic Sound Processing and Analysis/Synthesis” with Prof. Marlon Schumacher. The idea of this text is to show the technical solutions for my creative ideas and to share the knowledge gained to help the reader with their ideas. The purpose of this patch is to take sounds from everyday life and transform them into your own composition using several processes within Open Music.

Responsible: Veronika Reutz Drobnić, winter semester 21/22

Introduction, Iteration 1

The initial idea of the piece was to transform everyday sounds, for example the sound of a kettle, into a different, processed sound by implementing technical solutions in Open Music. This patch processes and merges several files into one composition. There are three iterations of the patch that I worked on during the semester. I will describe them in chronological order.

The original idea for the patch came from musique concréte. I wanted to make a 2-minute piece from concrete sounds (not synthesized in Open Music, but recorded). This patch consists of three subpatches that are connected to the maquette in the main patch.

The main patch

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Acousmatic study by Zeno Lösch

This article is about the three iterations of an acousmatic study by Zeno Lösch, 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 deals with the basic conception, ideas, constructive iterations and the technical implementation with OpenMusic.

Responsible persons: Zeno Lösch, Master student Music Informatics at HfM Karlsruhe, 1st semester

 

Idea and concept

 

I got my inspiration for this study from the Freeze effect of GRM Tools.

This effect makes it possible to layer a sample and play it back at different speeds at the same time.

With this process you can create independent compositions, sound objects, sound structures and so on.

My idea is to program the same with Open Music.

For this I used the maquette and om-loops.

In the OpenMusicPatch you can find the different processes of layering the source material.

The source material is a “filtered” violin. This was created using the cross-synthesis process. This process of the source material was not created in Open Music.

Source material

 

Music cannot exist without time. Our perception connects the different sounds and seeks a connection. In this process, also comparable to rhythm, the individual object is connected to other objects. Digital sound manipulation makes it possible to use processes to create other sounds from one sound, which are related to the same sound.

For example, I present the sound in one form and change it at another point in the composition. This usually creates a connection, provided the listener can understand it.

You can change a transposition or the pitch in a similar way to notes.

This changes the frequency of a note. With digital material, this can lead to very exciting results. On a piano, the overtones of each note are related to the fundamental. These are fixed and cannot be changed with traditional sheet music.

With digital material, the effect that transposes plays a very important role. Depending on the type of effect, I have various possibilities to manipulate the material according to my own rules.

The disadvantage with instruments is that with a violin, for example, the player can only play the note once. Ten times the same note means ten violins.

In OpenMusic it is possible to play the “instrument” any number of times (as long as the computer’s processing power allows it).

 

Process

To recreate the Grm-Freeze, a moquette was first filled with empty patches.

Filling a moquette with empty patches

 

The soundfile was then rendered from the moquette with an om-loop to the positions of the empty patches.

Loop for soundfile positions

 

The following code was used to avoid clipping.

Sox-Mix and Anti Clip

 

Layer Study first iteration

 

The source material is presented at the beginning. In the course of the study, it is repeatedly changed and stacked in different ways.

The study itself also plays with the dynamics. Depending on the sound stacking algorithm, the dynamics in each sound object are changed. As there is more than one sound in time, these sounds are normalized depending on how many sounds are present in the algorithm to avoid clipping.

The study begins with the source material. This is then presented in a different temporal sequence.

This layer is then filtered and is also quieter. The next one develops into a “reverberant” sound. A continuum. The continuum remains it is presented differently again.

In the penultimate sound, a form of glissandi can be heard, which again ends in a sound that is similar to the second, but louder.

The process of stacking and changing the sound is very similar for each section.

The position is given by the empty patch in the moquette.

Then the y-position and x-position parameters are used for modulation

Implementation of the x and y positions as modulation parameters
Layer study first iteration

 

Layer Study second iteration

I tried to create a different stereo image for each section.

Different rooms were simulated.

One technique that was used is the mid/side.

In this technique, the mid and side are extracted from a stereo signal using the following process:

Mid = (L R) * 0.5

Side = (L – R) * 0.5

An aural exciter has also been added.

In this process, the signal is filtered with a high-pass filter, distorted and added back to the input signal. This allows better definition to be achieved.

Through the mid/side, the aural exciter is only applied to one of the two and it is perceived as more “defined”.

To return the process to a stereo signal, the following process is used:

L = Mid Side

R = Mid – Side

Mid Side process

 

To further spatialize the sound, an all-pass filter and a comb filter were used to change the phase of the mid or side component.

Decorrelation of the phase

 

Layer Study Stereo

 

Layer Study third iteration

In this iteration, the stereo file was divided into eight speakers.

The different sections of the stereo composition were extracted and different splitting techniques were used.

In one of these, a different fade in and fade out was used for each channel.

In an acousmatic version of a composition, this fade in and fade out can be achieved with the controls of a mixer.

A mapcar and repeat-n were used for this purpose.

Random fades for multichannel

The position of the respective channels was changed in the other processes. A delay was used.

Multichannel delay

The final version is available on 2 channels.

Downmix Layer Study 8 channels to 2 channels