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Publisher Copyright: Author Feedback Delay Networks are one of the most popular and efficient means of generating artificial reverberation. Recently, we proposed the Grouped Feedback Delay Network (GFDN), which couples multiple FDNs while maintaining system stability. The GFDN can be used to model reverberation in coupled spaces that exhibit multi-stage decay. The block feedback matrix determines the inter- and intra-group coupling. In this article, we expand on the design of the block feedback matrix to include frequency-dependent coupling among the various FDN groups. We show how paraunitary feedback matrices can be designed to emulate diffraction at the aperture connecting rooms. Several methods for the construction of nearly paraunitary matrices are investigated. The proposed method supports the efficient rendering of virtual acoustics for complex room topologies in games and XR applications.

The cross-correlation of multichannel reverberation generated using interleaved velvet noise is studied. The interleaved velvet-noise reverberator was proposed recently for synthesizing the late reverb of an acoustic space. In addition to providing a computationally efficient structure and a perceptually smooth response, the interleaving method allows combining its independent branch outputs in different permutations, which are all equally smooth and flutter-free. For instance, a four-branch output can be combined in 4! or 24 ways. Additionally, each branch output set is mixed orthogonally, which increases the number of permutations from M! to M^2!, since sign inversions are taken along. Using specific matrices for this operation, which change the sign of velvet-noise sequences, decreases the correlation of some of the combinations. This paper shows that many selections of permutations offer a set of well decorrelated output channels, which produce a diffuse and colorless sound field, which is validated with spatial variation. The results of this work can be applied in the design of computationally efficient multichannel reverberators.

Flutter echo is a well-known acoustic phenomenon that occurs when sound waves bounce between two parallel reflective surfaces, creating a repetitive sound. In this work, we introduce a method to recreate flutter echo as an audio effect. The proposed algorithm is based on a feedback structure utilizing velvet noise that aims to synthesize the fluttery components of a reference room impulse response presenting flutter echo. Among these, the repetition time defines the length of the delay line in a feedback filter. The specific spectral properties of the flutter are obtained with a bandpass attenuation filter and a ripple filter, which enhances the harmonic behavior of the sound. Additional temporal shaping of a velvet-noise filter, which processes the output of the feedback loop, is performed based on the properties of the reference flutter. The comparison between synthetic and measured flutter echo impulse responses shows good agreement in terms of both the repetition time and reverberation time values.

The article is devoted to the systematization of the main modern methods of using reverberation in sound engineering and methodological bases of its use in the educational process in higher education institutions. Scientific novelty is to clarify the classification and systematization of the main modern methods of using artificial reverberation, as well as methodological principles of mastering artificial reverberation in the process of training. The purpose of the article is to systematize the main modern methods of using reverberation in sound engineering. To achieve this goal, the following methods were used: system-analytical, art history and comparative-historical. The main results and conclusions of the research. The evolution of digital technologies in the field of sound engineering, the introduction of reverberation algorithms have greatly contributed to the expansion of the use of artificial reverberation in sound engineering. The article briefly describes "convolutional" reverberation as one that operates with the most modern reverberation algorithm. Currently, a variety of effective methods of artificial reverberation using has been developed in sound engineering, which can be used to implement various creative tasks. The article describes the methods of using artificial reverberation for sound perspective creating; for back-vocal processing; for delay processing; for "softening" the sound of individual parts of musical instruments; for different signals of the same type processing; for solo batch processing using two reverb effects. Some types of artificial reverberation were used at different stages of the music industry development. The current state of software development allows the use of several reverb algorithms simultaneously, making it possible to create original sounds. In the process of mastering of various methods of using artificial reverberation by higher education students, it is best to start their acquaintance with artificial reverberation from the simplest reverberation effects, which can be implemented in "analog" effects processor, in which only some parameters vary based on the manufacturer's algorithms. Further mastering of artificial reverberation should be carried out through software (in particular, free VST-plugins of artificial reverberation) used in digital audio workstations (DAW)., Стаття присвячена систематизації основних сучасних методів використання реверберації у звукорежисурі та методичним основам її застосування у освітньому процесі у закладах вищої освіти. Наукова новизна полягає в уточненні класифікації та систематизації основних сучасних методів використання штучної реверберації, а також методичних засад опанування штучної реверберації у процесі професійної підготовки. Мета статті полягає у систематизації основних сучасних методів використання реверберації у звукорежисурі. Для реалізації мети використано методи: системно-аналітичний, мистецтвознавчий та порівняльно-історичний. Основні результати і висновки дослідження. Удосконалення цифрових технологій у галузі звукорежисури, запровадження ревербераційних алгоритмів значною мірою сприяли розширенню використання штучної реверберації у звукорежисурі. У статті коротко схарактеризовано «згорткову» реверберацію як таку, яка оперує найбільш сучасним ревербераційним алгоритмом. Наразі у звукорежисурі напрацьовано різноманітні ефективні методи використання штучної реверберації, які можуть застосовуватися для реалізації різноманітних творчих завдань. У статті схарактеризовано методи використання штучної реверберації для створення звукової перспективи; для обробки back-вокалу; для обробки дилею; для «пом’якшення» звучання окремих партій музичних інструментів; для обробки різних однотипних сигналів; для обробки партії соло з використанням двох ефектів реверберації. Окремі види штучної реверберації використовувалися на різних історичних етапах розвитку музичної індустрії. Сучасний стан розвитку програмного забезпечення дозволяє використовувати одразу декілька алгоритмів реверберації, завдяки чому наявна можливість створення оригінальних звучань. Упроцесі опанування здобувачами вищої освіти різних методів використання штучної реверберації найбільш доцільно розпочати їх знайомство зі штучною реверберацією з найпростіших ефектів реверберації, які можуть бути реалізовані в «аналоговому» процесорі ефектів, у якому на основі алгоритмів виробника варіюються лише деякі параметри. Подальше опанування штучної реверберації доцільно здійснювати через програмне забезпечення (зокрема, безкоштовні VST-плагіни штучної реверберації), що використовується у цифрових звукових робочих станціях (DAW).

The purpose of the article is to systematise the main modern methods of using reverberation in sound engineering. To achieve the goal, the following methods were used: systematic and analytical, art studies, comparative and historical. The scientific novelty consists in clarifying the classification and systematisation of the main modern methods of using artificial reverberation, as well as mastering the methodological foundations of artificial reverberation in the process of professional training. The main findings and conclusions of the study. The evolution of digital technologies in the field of sound engineering, the introduction of reverb algorithms contributed to the active use of artificial reverberation, in particular “convolution” reverb with the most modern reverb algorithm. The article considers artificial reverberation techniques for creating a sound perspective; for processing back vocals and delay effect; for "softening" the sound of certain parts of musical instruments; for processing various signals of the same type; for processing a solo part with two reverb effects. Certain types of artificial reverberation were used at different stages of the music industry development. The current state of software development makes it possible to use several reverberation algorithms at once and create an original sound. It is advisable for higher education students to start learning the artificial reverberation techniques with the simplest effects with a slight variation of only certain parameters. Further mastering of artificial reverberation should be carried out using software tools (in particular, free VST plug-ins) for digital audio workstations.

The purpose of the article is to systematise the main modern methods of using reverberation in sound engineering. To achieve the goal, the following methods were used: systematic and analytical, art studies, comparative and historical. The scientific novelty consists in clarifying the classification and systematisation of the main modern methods of using artificial reverberation, as well as mastering the methodological foundations of artificial reverberation in the process of professional training. The main findings and conclusions of the study. The evolution of digital technologies in the field of sound engineering, the introduction of reverb algorithms contributed to the active use of artificial reverberation, in particular “convolution” reverb with the most modern reverb algorithm. The article considers artificial reverberation techniques for creating a sound perspective; for processing back vocals and delay effect; for "softening" the sound of certain parts of musical instruments; for processing various signals of the same type; for processing a solo part with two reverb effects. Certain types of artificial reverberation were used at different stages of the music industry development. The current state of software development makes it possible to use several reverberation algorithms at once and create an original sound. It is advisable for higher education students to start learning the artificial reverberation techniques with the simplest effects with a slight variation of only certain parameters. Further mastering of artificial reverberation should be carried out using software tools (in particular, free VST plug-ins) for digital audio workstations., Цель статьи заключается в систематизации основных современных приемов использования реверберации в звукорежиссуре. Для реализации цели использованы методы: системно-аналитический, искусствоведческий и сравнительно-исторический. Научная новизна заключается в уточнении классификации и систематизации основных современных приемов использования искусственной реверберации, а также овладении методических основ искусственной реверберации в процессе профессиональной подготовки. Основные результаты и выводы исследования. Эволюция цифровых технологий в области звукорежиссуры, введение реверберационных алгоритмов способствовали активному использованию искусственной реверберации, в частности «свёрточной» реверберации с современным реверберационным алгоритмом. В статье рассмотрены приемы искусственной реверберации для создания звуковой перспективы; для обработки back-вокала и дилея; для «смягчения» звучания определенных партий музыкальных инструментов; для обработки различных однотипных сигналов; для обработки партии соло с двумя эффектами реверберации. Определенные разновидности искусственной реверберации использовались на различных этапах развития музыкальной индустрии. Современное состояние развития программного обеспечения позволяет использовать сразу несколько алгоритмов реверберации и создавать оригинальное звучание. Усвоения соискателями высшего образования приемов искусственной реверберации целесообразно начать с самых простых эффектов с незначительным варьированием определенных параметров. Дальнейшее освоение искусственной реверберации следует осуществлять с использованием средств программного обеспечения (в частности, при помощи бесплатных VST-плагинов) для цифровых звуковых рабочих станций., Мета статті полягає в систематизації основних сучасних прийомів використання реверберації у звукорежисурі. Для реалізації мети використано методи: системно-аналітичний, мистецтвознавчий та порівняльно-історичний. Наукова новизна полягає в уточненні класифікації та систематизації основних сучасних прийомів використання штучної реверберації, а також опануванні методичних засад штучної реверберації в процесі професійної підготовки. Основні результати і висновки дослідження. Еволюція цифрових технологій у галузі звукорежисури, запровадження ревербераційних алгоритмів сприяли активному використанню штучної реверберації, зокрема «згорткової» реверберації із найсучаснішим ревербераційним алгоритмом. У статті розглянуто прийоми штучної реверберації для створення звукової перспективи; для оброблення back-вокалу та дилею; для «пом’якшення» звучання певних партій музичних інструментів; для опрацювання різних однотипних сигналів; для оброблення партії соло з двома ефектами реверберації. Певні різновиди штучної реверберації використовувалися на різних етапах розвитку музичної індустрії. Сучасний стан розвитку програмного забезпечення дає змогу використовувати відразу декілька алгоритмів реверберації і створювати оригінальне звучання. Засвоєння здобувачами вищої освіти прийомів штучної реверберації доцільно розпочати з найпростіших ефектів із незначним варіюванням лише певних параметрів. Подальше опанування штучної реверберації варто здійснювати засобами програмного забезпечення (зокрема, з допомогою безплатних VST-плагінів) для цифрових звукових робочих станцій.

This document is an annex of the following paper: Achille Aknin and Roland Badeau.Stochastic reverberation model with a frequency dependent attenuation.IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2021 (submitted for publica-tion).It provides mathematical proofs of theorems that we use in the paper. More specifically, we will prove that the set P, of which we recall the definition in Section 1, is a matrix Lie group in Section 2 and that we can swap matrices of P with lower triangular Toeplitz matrices in Section 3.; International audience; In various audio signal processing applications, such as source separation and dereverberation, accurate mathematical modeling of both source signals and room reverberation is needed to properly describe the audio data. In a previous paper, we introduced a stochastic room impulse response model based on the image source principle, and we proposed an expectation-maximization algorithm that was able to efficiently estimate the model parameters in various experimental settings. This paper aims to extend the model in order to account for the dependency of the exponential decay over frequency, due to the walls usually absorbing less energy at low frequencies than at high frequencies. Our experimental results show that this refinement of the model is able to generate realistic room impulse responses, that are perceptively very close to the original ones.

Reverberation is one of the most important effects used in audio production. Although nowadays numerous real-time implementations of artificial reverberation algorithms are available, many of them depend on a database of recorded or pre-synthesized room impulse responses, which are convolved with the input signal. Implementations that use an algorithmic approach are more flexible but do not let the users have full control over the produced sound, allowing only a few selected parameters to be altered. The real-time implementation of an artificial reverberation synthesizer presented in this study introduces an audio plugin based on a feedback delay network (FDN), which lets the user have full and detailed insight into the produced reverb. It allows for control of reverberation time in ten octave bands, simultaneously allowing adjusting the feedback matrix type and delay-line lengths. The proposed plugin explores various FDN setups, showing that the lowest useful order for high-quality sound is 16, and that in the case of a Householder matrix the implementation strongly affects the resulting reverberation. Experimenting with delay lengths and distribution demonstrates that choosing too wide or too narrow a length range is disadvantageous to the synthesized sound quality. The study also discusses CPU usage for different FDN orders and plugin states.

Feedback delay networks (FDNs) are recursive filters, which are widely used for artificial reverberation and decorrelation. One central challenge in the design of FDNs is the generation of sufficient echo density in the impulse response without compromising the computational efficiency. In a previous contribution, we have demonstrated that the echo density of an FDN can be increased by introducing so-called delay feedback matrices where each matrix entry is a scalar gain and a delay. In this contribution, we generalize the feedback matrix to arbitrary lossless filter feedback matrices (FFMs). As a special case, we propose the velvet feedback matrix, which can create dense impulse responses at a minimal computational cost. Further, FFMs can be used to emulate the scattering effects of non-specular reflections. We demonstrate the effectiveness of FFMs in terms of echo density and modal distribution.

A recent publication introduced the Directional Feedback Delay Network, a parametric artificial reverberation algorithm capable of producing direction-dependent energy decay. This method extends the capabilities of Feedback Delay Networks by using multichannel delay-line groups and a spatial transform to produce direction-dependent reverberation in the Ambisonics domain. In this paper, we present a modified formulation of the Directional Feedback Delay Network method that allows both frequency-and direction-dependent reverberation. Multichannel delay-line groups are used to manipulate signals incident on a spherical grid through independent recursive signal paths, while an early reflection module gives a physically-motivated spatial distribution of input signals in the system. The overall number of delay lines is reduced from the previous formulation, and the design allows flexibility to favor a lower computational cost over accuracy. acceptedVersion

Feedback delay networks (FDNs) belong to a general class of recursive filters which are widely used in sound synthesis and physical modeling applications. We present a numerical technique to compute the modal decomposition of the FDN transfer function. The proposed pole finding algorithm is based on the Ehrlich-Aberth iteration for matrix polynomials and has improved computational performance of up to three orders of magnitude compared to a scalar polynomial root finder. The computational performance is further improved by bounds on the pole location and an approximate iteration step. We demonstrate how explicit knowledge of the FDN's modal behavior facilitates analysis and improvements for artificial reverberation. The statistical distribution of mode frequency and residue magnitudes demonstrate that relatively few modes contribute a large portion of impulse response energy.

International audience; Le Spatialisateur est une suite logicielle développée à l’Ircam et déstinée à la spatialisation sonore et à la réverbération artificielle. Cet article dresse un bilan des développements réalisés au cours des dernières années.

Virtual Reality (VR) aims at the creation of responsive simulations, that provide humans the illusion of a world or environment, they can interact with. Therefore, the user is stimulated with sensory cues that are computer-generated, based on a model of a virtual world (scene). Considering the sense of hearing, the acoustic description of the scene is transformed into auditory stimuli, which are then provided using headphones or loudspeakers.Signal processing is fundamental to this process, called auralization. It involves digital filtering in several uses and in diverse forms (e.g. non-linear and linear filtering, time-invariant and time-varying filtering). A common requirement for VR is a low latency (immediate system response). The computational extent however, ranges from moderate to highly complex, depending on the application.This work focuses on finite impulse response filters (FIR filters), which are applied in binaural synthesis, spatial sound reproduction and artificial reverberation. Straightforward FIR filtering in the time-domain fails to satisfy the requirements stated above. These are met by implementing the FIR filtering using efficient mathematical algorithms for fast convolution. Since the 1960s different algorithmic concepts have been developed, often from the divide-and-conquer paradigm. The most popular example is fast convolution using the fast Fourier transform (FFT), which established as the standard tool. However, also fast convolution algorithms must be adapted to serve for real-time filtering. The most powerful concept hereby is partitioned convolution, which first splits the operands and then solves the partial problems using a fast convolution technique. Essential is that the decomposition conforms with real-time processing. This thesis considers three different classes of partitioned convolution algorithms for the use in real-time auralization: uniformly and non-uniformly partitioned filters, as well as unpartitioned filters. The algorithmic properties of each class are derived and guidelines for an optimal choice of parameters are provided. All techniques are analyzed regarding multi-channel processing, networks of filters and time-varying filtering, as needed in Virtual Reality. The work identifies suitable convolution techniques for different applications, ranging from resource-aware auralization on mobile devices to extensive room acoustical auralization on dedicated multi-processor systems.

Artificial reverberation is usually introduced, as a digital audio effect, to give a sense of enclosing architectural space. In this paper we argue about the effectiveness and usefulness of diffusive reverberators in physically-inspired sound synthesis. Examples are given for the synthesis of textural sounds, as they emerge from solid mechanical interactions, as well as from aerodynamic and liquid phenomena.

International audience; Ircam Spat is a software suite dedicated to sound spatialization and artificial reverberation. This paper traces the evolution of the software during the last few years and presents its current state with a focus on the newly available features.

cote interne IRCAM: Jot99a; None / None; National audience; This paper overviews the principles and methods for synthesizing complex three-dimensional sound scenes by processing multiple individual source signals. Signal processing techniques for directional sound encoding and rendering over loudspeakers or headphones are reviewed, as well as algorithms and interface models for synthesizing and dynamically controlling room reverberation and distance effects. A real-time modular spatial sound processing software system, called Spat, is presented. It allows reproducing and controlling the localization of sound sources in three dimensions and the reverberation of sounds in an existing or virtual space. A particular aim of the Spatialisateur project is to provide direct and computationally efficient control over perceptually relevant parameters describing the interaction of each sound source with the virtual space, irrespective of the chosen reproduction format over loudspeakers or headphones. The advantages of this approach are illustrated in practical contexts, including professional audio, computer music, multimodal immersive simulation systems, and architectural acoustics.

It is shown that half of the points on the diamond grid are redundant for sampling and reconstructing a bandlimited 3-D signal. This redundancy is then exploited to show that the tetrahedral digital waveguide mesh requires four times more computational density and twice the memory storage for the same approximation as a finite difference scheme on the face-centered cubic lattice.

The digital emulation of analog audio effects and synthesis components, through the simulation of lumped circuit components has seen a large amount of activity in recent years; electromechanical effects have seen rather less, primarily because they employ distributed mechanical components, which are not easily dealt with in a rigorous manner using typical audio processing constructs such as delay lines and digital filters. Spring reverberation is an example of such a system--a spring exhibits complex, highly dispersive behavior, including coupling between different types of wave propagation (longitudinal and transverse). Standard numerical techniques, such as finite difference schemes are a good match to such a problem, but require specialized design and analysis techniques in the context of audio processing. A model of helical spring vibration is introduced, along with a family of finite difference schemes suitable for time domain simulation. Various topics are covered, including numerical stability conditions, tuning of the scheme to the response of the model system, numerical boundary conditions and connection to an excitation and readout, implementation details, as well as computational requirements. Simulation results are presented, and full energy-based stability analysis appears in an Appendix.

cote interne IRCAM: Jot97b; None / None; National audience; This paper discusses efficient digital signal processing algorithms for real-time synthesis of dynamically controllable, natural-sounding artificial reverberation. A general modular framework is proposed for configuring a spatial sound processing and mixing system according to the reproduction format or setup and the listening conditions, over loudspeakers or headphones. In conclusion, the implementation and applications of a spatial sound processing software are described, and approaches to control interface design and effective. distance effects are reviewed.