Category Archive Symbolische Klangverarbeitung und Analyse/Synthese

ByLukas Körfer

Wave field synthesis with OM-SoX

Abstract: This final project was created at the end of the winter semester 2023/24 as part of the course “Symbolische Klangverarbeitung und Analyse/Synthese” (eng. Symbolic Sound Processing and Analysis/Synthesis) of the MA Music Informatics. An application for sound spatialization was developed in the program OpenMusic using the library OM-SoX implementing Steinberg and Snow’s “acoustic curtain”, a technique for wave field synthesis.

Responsible: Lukas Körfer

Wave field synthesis

Wave field synthesis (WFS for short) is the spatialization of virtual sound sources using a high-density loudspeaker array. This spatialization technique attempts to reproduce a physical soundfield over an extended area in a way to provide for multiple non-conicident listening positions a congruent impression of the localization of sound sources. This is achieved by generating a wave field consisting of a large number of individual sound sources that are synchronized in such a way that a coherent sound wave is created, for which given certain constraints it should be possible to localize a virtual sound source in the room.

 

For a better understanding of how WFS works, the subject can be approached via the physical phenomenon of interference pattern formation behind an obstacle with openings. When a wave encounters one or more slits, it is diffracted through the openings and propagates behind the obstacle. This leads to the formation of a pattern of wave interference on the other side of the obstacle. Similarly, wave field synthesis uses an array of loudspeakers to generate a coherent sound wave. This requires precise calculation and control of the phase and amplitude relationships of the sound waves emanating from each speaker. These calculations are dependent on the distances of each individual loudspeaker in the array relative to the position in space of the respective virtual sound source.

Project description

For this project, a program was to be created with the general goal of ultimately obtaining a multi-channel audio file that can be used for wave field synthesis with a loudspeaker array through certain influence and adjustments by a user. To achieve this, it was first necessary to design which parameters should be set and influenced by the user of the program.

User input

 

In addition to the audio file, which is to be used for spatialization, the user must specify certain information about the loudspeaker array on the one hand and the position or positions of one or more virtual sound sources relative to the loudspeaker array on the other. In order to make the configuration of the program as simple and intuitive as possible, I have decided to mainly use a picture object in which the structure can be recorded. The positions of the loudspeakers can be specified by drawing a rectangle and those of the virtual sound sources with circles. One or more circles can be drawn, with each circle representing a sound source. The loudspeakers can be specified in two different ways. If only a single rectangle is drawn in the picture object, this represents the area of a loudspeaker array. In order to be able to determine the specific positions of the individual loudspeakers in the next step of the program, two additional pieces of information are required. Firstly, the length of the loudspeaker array in meters; this also influences the scale for the complete drawn setup. Secondly, the number of loudspeakers in the drawn area must be specified. As soon as more than one rectangle is specified by the user, each individual rectangle represents an individual loudspeaker. In order to be able to specify a scale for the drawn structure in this variant – which was previously possible by specifying the length of the loudspeaker array – the width/height of the area of the complete picture object can now be specified. The first variant, where the loudspeaker array can only be drawn with a rectangle, makes the application much less complicated, but also requires the loudspeakers to be linear and evenly spaced.

Calculating distances

 

Once all the graphics of the picture object have been read out, they must be divided into rectangles and circles for further processing. If only one rectangle is found, the position and dimension of the rectangle and the two specifications for the length and number of loudspeaker arrays can first be used to determine the position of each individual loudspeaker within the array in meters. If there are several rectangles, this step is not necessary and the center points of all specified rectangles are simply determined. It is then possible to calculate the Euclidean distance from all sources to each individual loudspeaker on the same scale using another Lisp function. It should be noted that all graphics drawn by the user in the Picture object that do not correspond to a rectangle or a circle are ignored and not taken into account for the further calculations. As any number of virtual sound sources can be specified for the application, all circles that exist in the picture object are also captured in this step, whereby the order is irrelevant.

Sound processing

 

Sound processing is implemented in the next section of the program. Basically, a multi-channel file is created with the sound file specified by the user together with the previously calculated distances, which can be used for the intended loudspeaker array. This process takes place in a nested OM loop with two levels.

 

In the first level, it is first iterated over each element within the distance list. Each of these elements corresponds to a list that belongs to a virtual sound source, which contains the distances to each loudspeaker. Before the process enters the second level of the loop, further calculations are performed in a Lisp function using the current distance list.

This function iterates over each distance and determines the time delay, volume reduction and a cutoff frequency for a lowpass filter to calculate the air absorption of high frequencies and collects them in a list. In the next step, the result of this Lisp function is used to enter the second level of the loop.

 

Here, the respective SoX effect is applied to the calculated value; SoX level for volume reduction, SoX lowpass for air absorption and SoX pad for the time delay. The resulting audio file is saved for each iteration. Each of the three lists has as many values as the previously calculated distances from the current sound source to the speakers. This means that each audio file saved in this loop represents one channel of the subsequent multi-channel file for the current sound source.

The multi-channel file can now be created in the next step in the first layer with SoX-Merge and stored temporarily at the end of the loop. This process is repeated for all remaining virtual sound sources (if existing) and are collected as the output of this upper loop. All multi-channel files of the respective sound sources are then merged with a SoX-Mix.

If only one virtual sound source is specified by the user, the output of the outermost loop will only consist of a single multi-channel file for this one source. In this case, the SoX-Mix is not required and it would even lead to an error during the evaluation of the program if the input of the SoX-Mix consisted of only one audio file. The OM-If therefore avoids the use of the SoX-Mix as soon as the output of the patcher, in which the distances are determined, only consists of one list, which means that only one circle for a virtual sound source has been drawn in the picture object.

Finally, silence can be added to the multi-channel file using the SoX pad, depending on preference, if the selected audio file is particularly short, for example. At the same time, the final multi-channel file is saved in Outfile as “wfsOutFile.wav”.

ByFlorian Simon

Interspaces – Acousmatic study with OM-SoX

Interspaces juxtaposes sounds from human civilization with sounds from nature. Four pairs of field recordings are presented, which are filtered according to the principle of a vocoder according to the spectrum of a section of the counterpart.

Responsible: Florian Simon

Interspaces shows the following four pairs (format: total recording – source of the spectrum):

  1. Chirping Arctic terns – Vowel “E” called by humans
    Lively market, people talking and calling – Arctic tern call

  2. Rippling of a river – Accelerating car
    Main road – rushing of a river

  3. Forest scenery, rustling leaves and birds – Train horn
    Station concourse – chirping of a songbird

  4. Thunderstorm – clinking of cutlery
    Business in a restaurant kitchen – thunder

The field recordings come from the FreeToUseSounds library.

Interspaces uses an equilateral octagonal loudspeaker arrangement, whereby the two channels of the source material are each placed at opposite points in the array. The two recordings of a pair are also offset by 90 degrees from each other by default, so that four sound sources can be perceived.

Each recording is divided into several sections of random size within a certain frame and concatenated again in randomized order with short crossfades. The number of sections increases with each pair of recordings: 4, 9, 16 and finally 23. With each new section, the two sound sources also “move” in the array by 0.25 channels in a certain direction. Since the number of sections is the same for both recordings of a pair, but not the position of the cuts, deviations from the base of a 90-degree spacing and a greater variety of sounds are created. Interspaces is designed as an installation to allow free exploration of the stereo fields.

Interspaces was created in OpenMusic using functions from the OM-SoX library. The underlying program consists of two parts. The first is used to create the manipulated recordings by spectral analysis (sox-dft), splitting the source material into up to 4096 frequency bands (sox-sinc), adjusting their volume levels according to the generated spectrum (sox-level) and reassembling them (sox-mix).

The second part of the program uses the synthesis patch of a maquette to control the division into sections (sox-trim) and their spatialization (sox-remix) and final alignment (sox-splice) for each of the eight generated audio files, and finally to organize the finished blocks in terms of time (sox-pad and sox-mix). In the last step, the time saved by the crossfades must be taken into account and subtracted from the onset value/x position in the maquette.

Audio (binaural mixed to stereo):

Alex Player - Best audio player
 

Unfortunately, this vocoder method has the disadvantage that the individual frequency bands are initially very quiet and therefore artefacts in the form of noise occur when applying the gain and the final normalization. Conversely, clipping occurs when certain frequencies are strongly represented in both source recordings. If you lower the gain values accordingly to avoid this, quieter sections in the result may be barely audible, depending on the size of the dynamic difference. The noise can be easily eliminated by selecting higher gain values, but this increases the clipping problem. In the above version of Interspaces, the best compromise between the two effects was sought for all eight audio clips.


 

 

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

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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: