Browsing by Author "Schlecht, Sebastian J."

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Acoustic analysis and dataset of transitions between coupled rooms
(2021) McKenzie, Thomas; Schlecht, Sebastian J.; Pulkki, Ville
A4 Artikkeli konferenssijulkaisussa
The measurement of room acoustics plays a wide role in audio research, from physical acoustics modelling and virtual reality applications to speech enhancement. While vast literature exists on position-dependent room acoustics and coupling of rooms, little has explored the transition from one room to its neighbour. This paper presents the measurement and analysis of a dataset of spatial room impulse responses for the transition between four coupled room pairs. Each transition consists of 101 impulse responses recorded using a fourth-order spherical microphone array in 5 cm intervals, both with and without a continuous line-of-sight between the source and microphone. A numerical analysis of the room transitions is then presented, including direct-to-reverberant ratio and direction of arrival estimations, along with potential applications and uses of the dataset.
Allpass Feedback Delay Networks
(2021-01-21) Schlecht, Sebastian J.
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
In the 1960s, Schroeder and Logan introduced delay line-based allpass filters, which are still popular due to their computational efficiency and versatile applicability in artificial reverberation, decorrelation, and dispersive system design. In this work, we extend the theory of allpass systems to any arbitrary connection of delay lines, namely feedback delay networks (FDNs). We present a characterization of uniallpass FDNs, i.e., FDNs, which are allpass for an arbitrary choice of delays. Further, we develop a solution to the completion problem, i.e., given an FDN feedback matrix to determine the remaining gain parameters such that the FDN is allpass. Particularly useful for the completion problem are feedback matrices, which yield a homogeneous decay of all system modes. Finally, we apply the uniallpass characterization to previous FDN designs, namely, Schroeder's series allpass and Gardner's nested allpass for single-input, single-output systems, and, Poletti's unitary reverberator for multi-input, multi-output systems and demonstrate the significant extension of the design space.
Audio peak reduction using a synced allpass filter
(2022) Schlecht, Sebastian J.; Fierro, Leonardo; Välimäki, Vesa; Backman, Juha
A4 Artikkeli konferenssijulkaisussa
Peak reduction is a common step used in audio playback chains to increase the loudness of a sound. The distortion introduced by a conventional nonlinear compressor can be avoided with the use of an allpass filter, which provides peak reduction by acting on the signal phase. This way, the signal energy around a waveform peak can be smeared while maintaining the total energy of the signal. In this paper, a new technique for linear peak amplitude reduction is proposed based on a Schroeder allpass filter, whose delay line and gain parameters are synced to match peaks of the signal's autocorrelation function. The proposed method is compared with a previous search method and is shown to be often superior. An evaluation conducted over a variety of test signals indicates that the achieved peak reduction spans from 0 to 5 dB depending on the input waveform. The proposed method is widely applicable to real-time sound reproduction with a minimal computational processing budget.
The auditory perceived aperture position of the transition between rooms
(2022-09-01) McKenzie, Thomas; Schlecht, Sebastian J.; Pulkki, Ville
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
This exploratory study investigates the phenomenon of the auditory perceived aperture position (APAP): the point at which one feels they are in the boundary between two adjoined spaces, judged only using auditory senses. The APAP is likely the combined perception of multiple simultaneous auditory cue changes, such as energy, reverberation time, envelopment, decay slope shape, and the direction, amplitude, and colouration of direct and reverberant sound arrivals. A framework for a rendering-free listening test is presented and conducted in situ, avoiding possible inaccuracies from acoustic simulations, impulse response measurements, and auralisation to assess how close the APAP is to the physical aperture position under blindfold conditions, for multiple source positions and two room pairs. Results indicate that the APAP is generally within ± 1 m of the physical aperture position, though reverberation amount, listener orientation, and source position affect precision. Comparison to objective metrics suggests that the APAP generally falls within the period of greatest acoustical change. This study illustrates the non-trivial nature of acoustical room transitions and the detail required for their plausible reproduction in dynamic rendering and game audio engines.
Auralisation of the Transition between Coupled Rooms
(2021-09-10) McKenzie, Thomas; Schlecht, Sebastian J.; Pulkki, Ville
A4 Artikkeli konferenssijulkaisussa
The perceptual experience of the transition between coupled rooms remains a little investigated area of research. This paper presents a pipeline for auralising the transition between coupled rooms, utilising a time-varying partitioned convolution for fast position-dependent switching between spatial room impulse responses (SRIRs) and parametric binaural rendering over highly acoustically transparent headphones, with in-situ calibration to the corresponding real-world acoustics. The system is verified by an in-situ listening test with both real and virtual stimuli, conducted in six degrees-of-freedom virtual reality with three-dimensional visuals from measured room models. Results show that the auralisation is rated as highly natural, equalling the naturalness of the corresponding real world auditory stimuli. This pipeline is therefore appropriate for testing of coupled room transition algorithms and SRIR interpolation techniques, as well as non-in-situ testing.
Auralization of Measured Room Transitions in Virtual Reality
(2023-06-03) McKenzie, Thomas; Meyer-Kahlen, Nils; Hold, Christoph; Schlecht, Sebastian J.; Pulkki, Ville
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
To auralize a room’s acoustics in six degrees-of-freedom virtual reality (VR), a dense set of spatial room impulse response (SRIR) measurements is required, so interpolating between a sparse set is desirable. This paper studies the auralization of room transitions by proposing a baseline interpolation method for higher-order Ambisonic SRIRs and evaluating it in VR. The presented method is simple yet applicable to coupled rooms and room transitions. It is based on linear interpolation with RMS compensation, although direct sound, early reflections, and late reverberation are processed separately, whereby the input direct sounds are first steered to the relative direction-of-arrival before summation and interpolated early reflections are directionally equalized. The proposed method is first evaluated numerically, which demonstrates its improvements over a basic linear interpolation. A listening test is then conducted in six degrees-of-freedom VR, to assess the density of SRIR measurements needed in order to plausibly auralize a room transition using the presented interpolation method. The results suggest that, given the tested scenario, a 50-cm to 1-m inter-measurement distance can be perceptually sufficient.
Blind Directional Room Impulse Response Parameterization from Relative Transfer Functions
(2022) Meyer-Kahlen, Nils; Schlecht, Sebastian J.
A4 Artikkeli konferenssijulkaisussa
Acquiring information about an acoustic environment without conducting dedicated measurements is an important problem of forthcoming augmented reality applications, in which real and virtual sound sources are combined. We propose a straightforward method for estimating directional room impulse responses from running signals. We adaptively identify relative transfer functions between the output of a beam-former pointing into the direction of a single active sound source and the complete set of spherical harmonics domain signals, representing all directions. To this end, estimation is performed with a frequency domain recursive least squares algorithm. Then, parameters such as the directions of arrival of early reflections and the reverberation time are extracted. Estimation of the direct-to-reverberant ratio requires dedicated processing. We show examples of successful estimation from speech signals, based on a simulated and a measured response.
Bounded-Magnitude Discrete Fourier Transform [Tips & Tricks]
(2023-05-01) Schlecht, Sebastian J.; Valimaki, Vesa; Habets, Emanuel A.P.
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
Analyzing the magnitude response of a finite-length sequence is a ubiquitous task in signal processing. However, the discrete Fourier transform (DFT) provides only discrete sampling points of the response characteristic. This work introduces bounds on the magnitude response, which can be efficiently computed without additional zero padding. The proposed bounds can be used for more informative visualization and inform whether additional frequency resolution or zero padding is required.
Calibrating the Sabine and Eyring formulas
(2022-08-24) Prawda, Karolina; Schlecht, Sebastian J.; Välimäki, Vesa
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
Of the many available reverberation time prediction formulas, Sabine's and Eyring's equations are still widely used. The assumptions of homogeneity and isotropy of sound energy during the decay associated with those models are usually recognized as a reason for lack of agreement between predictions and measurements. At the same time, the inaccuracy in the estimation of the sound-absorption coefficient adds to the uncertainty of calculations. This paper shows that the error of incorrectly assumed sound absorption is more detrimental to the prediction precision than the inherent error in the formulas themselves. The proposed absorption calibration procedure reduces the differences between the measured and predicted reverberation time values, showing that an accuracy within ±10% from the target reverberation time values can be achieved regardless of the absorption distribution in a room. The paper also discusses the oft neglected air absorption of sound, which may introduce considerable bias to the measurement results. The need for an air-absorption compensation procedure is highlighted, and a method for the estimation of its parameters in octave bands is proposed and compared with other approaches. The results of this study provide justification for the use of the Sabine and Eyring formulas for reverberation time predictions.
Common-slope modeling of late reverberation
(2023) Gotz, Georg; Schlecht, Sebastian J.; Pulkki, Ville
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
The decaying sound field in rooms is typically described by energy decay functions (EDFs). Late reverberation can deviate considerably from the ideal diffuse field, for example, in multiple connected rooms or non-uniform absorption material distributions. This paper proposes the common-slope model of late reverberation. The model describes spatial and directional late reverberation as linear combinations of exponential decays called common slopes. Its fundamental idea is that common slopes have decay times that are invariant across space and direction, while their amplitudes vary across both. We explore different approaches for determining the common slopes for large EDF sets describing different source-receiver configurations of the same environment. Among the presented approaches, the k-means clustering of decay times is the most general. Our evaluation shows that the common-slope model introduces only a small error between the modeled and the true EDF, while being considerably more compact than the traditional multi-exponential model. The amplitude variations of the common slopes yield interpretable room acoustic analyses. The common-slope model has potential applications in all fields relying on late reverberation models, such as source separation, dereverberation, echo cancellation, and parametric spatial audio rendering.
Common-slope modeling of late reverberation in coupled rooms
(2022) Götz, Georg; Schlecht, Sebastian J.; Pulkki, Ville
A4 Artikkeli konferenssijulkaisussa
Coupled rooms have a distinct sound energy decay behavior, which exhibits more than one decay time under certain conditions. The sound energy decay analysis in such scenarios requires decay models consisting of multiple exponentials with distinct decay rates and amplitudes. While multi-exponential decay analysis is commonly used in room acoustics, the spatial and directional sound energy decay variations in coupled rooms have received little attention. In this work, we introduce the common-slope model of late reverberation for coupled rooms. Common slopes are spatially and directionally invariant decay functions over time, whose amplitudes model all decay variations with respect to the source-receiver configuration. For example, in a scene consisting of two coupled rooms, it is possible to determine two common decay times that approximate the decay for all source-receiver configurations in the scene. Consequently, all spatial and directional decay variations are expressed via decay amplitudes only. We apply the common-slope analysis to measurements of room transitions between coupled rooms. Our analysis shows that the common-slope model approximates the measured sound energy decay with little error. The proposed common-slope model can be used for room acoustic analysis and the efficient synthesis of artificial late reverberation tails.
Dark Velvet Noise
(2022) Fagerström, Jon; Meyer-Kahlen, Nils; Schlecht, Sebastian J.; Välimäki, Vesa
A4 Artikkeli konferenssijulkaisussa
This paper proposes dark velvet noise (DVN) as an extension of the original velvet noise with a lowpass spectrum. The lowpass spectrum is achieved by allowing each pulse in the sparse sequence to have a randomized pulse width. The cutoff frequency is controlled by the density of the sequence. The modulated pulse-width can be implemented efficiently utilizing a discrete set of recursive running-sum filters, one for each unique pulse width. DVN may be used in reverberation algorithms. Typical room reverberation has a frequency-dependent decay, where the high frequencies decay faster than the low ones. A similar effect is achieved by lowering the density and increasing the pulse-width of DVN in time, thereby making the DVN suitable for artificial reverberation.
A dataset of higher-order Ambisonic room impulse responses and 3D models measured in a room with varying furniture
(2021-09-23) Götz, Georg; Schlecht, Sebastian J.; Pulkki, Ville
A4 Artikkeli konferenssijulkaisussa
This paper presents Motus, a new dataset of higher-order Ambisonic room impulse responses. The measurements took place in a single room while varying the amount and placement of furniture. 830 different room configurations were measured with four source-to-receiver configurations, resulting in 3320 room impulse responses in total. The dataset features various furniture object placements, including non-uniform distributions of absorptive material and cases with occluded direct paths between source and receiver. All acoustic measurements are accompanied by matching 3D models and 360°-photographs of the room. After describing the dataset, we demonstrate its usage with a reverberation time analysis. The analysis reveals that most of our measurements follow the expected relationship between absorption area and reverberation time. Some exceptional cases feature particular room acoustic phenomena, such as non-uniform absorption area distributions or multi-slope decays. Additionally, we show with a large number of measurements that furniture placement can significantly affect the reverberation time of a room. The dataset can be used to investigate room acoustic topics such as the acoustic effects of absorber placements or the decay behavior of rooms.
Decorrelation in Feedback Delay Networks
(2023) Schlecht, Sebastian J.; Fagerström, Jon; Valimaki, Vesa
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
The feedback delay network (FDN) is a popular filter structure to generate artificial spatial reverberation. A common requirement for multichannel late reverberation is that the output signals are well decorrelated, as too high a correlation can lead to poor reproduction of source image and uncontrolled coloration. This article presents the analysis of multichannel correlation induced by FDNs. It is shown that the correlation depends primarily on the feedforward paths, while the long reverberation tail produced by the recursive path does not contribute to the inter-channel correlation. The impact of the feedback matrix type, size, and delays on the inter-channel correlation is demonstrated. The results show that small FDNs with a few feedback channels tend to have a high inter-channel correlation, and that the use of a filter feedback matrix significantly improves the decorrelation, often leading to the lowest inter-channel correlation among the tested cases. The learnings of this work support the practical design of multichannel artificial reverberators for immersive audio applications.
Directional distribution of the pseudo intensity vector in anisotropic late reverberation
(2024-02-01) Meyer-Kahlen, Nils; Schlecht, Sebastian J.
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
The pseudo intensity vector (PIV) is often used to analyze the directional properties of spatial room impulse responses. In the early part of the response, it is capable of estimating the directions of individual reflections. However, thus far, its behaviour in the late field is unclear. Specifically, it is unknown whether anisotropy, i.e., a direction-dependent energy distribution, is captured by the directional estimates. In this study, a closed-form expression of the directional distribution of the pressure-normalized pseudo intensity vector contingent on a general stochastical model of anisotropic fields was analytically derived. This paper shows that the probability density function of this PIV is a multivariate Cauchy distribution, which does indeed depend on the energy distribution of the field, yet the directional distribution has very limited degrees of freedom. The derived distribution is compared to the results of Monte Carlo simulations and fields captured with a microphone array in a real room. These results facilitate better understanding of the behaviour of parametric spatial room impulse response methods and may enable improved directional estimators for anisotropic fields.
Directional feedback delay network
(2019-01-01) Alary, Benoit; Politis, Archontis; Schlecht, Sebastian J.; Välimäki, Vesa
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
Artificial reverberation algorithms are used to enhance dry audio signals. Delay-based reverberators can produce a realistic effect at a reasonable computational cost. While the recent popularity of spatial audio algorithms is mainly related to the reproduction of the perceived direction of sound sources, there is also a need to spatialize the reverberant sound field. Usually multichannel reverberation algorithms output a series of decorrelated signals yielding an isotropic energy decay. This means that the reverberation time is uniform in all directions. However, the acoustics of physical spaces can exhibit more complex direction-dependent characteristics. This paper proposes a new method to control the directional distribution of energy over time, within a delay-based reverberator, capable of producing a directional impulse response with anisotropic energy decay. We present a method using multichannel delay lines in conjunction with a direction-dependent transform in the spherical harmonic domain to control the direction-dependent decay of the late reverberation. The new reverberator extends the feedback delay network, retaining its time-frequency domain characteristics. The proposed directional feedback delay network reverberator can produce non-uniform direction-dependent decay time, suitable for anisotropic decay reproduction on a loudspeaker array or in binaural playback through the use of ambisonics.
Distribution of Modal Damping in Absorptive Shoebox Rooms
(2023) Schafer, Maximilian; Prawda, Karolina; Rabenstein, Rudolf; Schlecht, Sebastian J.
A4 Artikkeli konferenssijulkaisussa
The image-source method is widely applied to compute room impulse responses (RIRs) of shoebox rooms with arbitrary damping. However, with increasing RIR lengths, the number of image sources grows rapidly, leading to slow computation. We propose a method to estimate the damping density of a damped shoebox room, which in turn can provide the energy decay necessary to model the stochastic late reverberation. The damping density is derived from a modal decomposition that is compliant with the ISM solution. We show that the proposed method gives a more accurate estimate of the energy decay than previous methods and can be efficiently computed regardless of the RIR lengths. While we focus on the derivation and evaluation, the main practical applications of the proposed model include, e.g., the faster synthesis of late reverb and the analysis of multi-slope decays.
Dynamic late reverberation rendering using the common-slope model
(2024-04-06) Götz, Georg; Kerimovs, Teodors; Schlecht, Sebastian J.; Pulkki, Ville
A4 Artikkeli konferenssijulkaisussa
Late reverberation rendering in video games and virtual reality applications can be challenging due to limited computational resources. Typical scenes feature complex geometries with multiple coupled rooms or non-uniform absorption. Additionally, the audio engine must continuously adapt to the player’s movements and the sound sources in the scene. This paper proposes a dynamic rendering system for anisotropic and inhomogeneous late reverberation. It is based on the common-slope model and uses a set of exponentially decaying reverberators that are weighted with position-, direction-, and frequency-dependent gains. We evaluate the system in a scene consisting of three coupled rooms, where we illustrate the reverberator gains for multiple octave bands. The proposed method allows real-time rendering of the spatial late reverberation while using a small number of artificial reverberators.
Generating coherence-constrained multisensor signals using balanced mixing and spectrally smooth filters
(2021-03-01) Mirabilii, Daniele; Schlecht, Sebastian J.; Habets, Emanuël A.P.
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
The spatial properties of a noise field can be described by a spatial coherence function. Synthetic multichannel noise signals exhibiting a specific spatial coherence can be generated by properly mixing a set of uncorrelated, possibly non-stationary, signals. The mixing matrix can be obtained by decomposing the spatial coherence matrix. As proposed in a widely used method, the factorization can be performed by means of a Choleski or eigenvalue decomposition. In this work, the limitations of these two methods are discussed and addressed. In particular, specific properties of the mixing matrix are analyzed, namely, the spectral smoothness and the mix balance. The first quantifies the mixing matrix-filters variation across frequency and the second quantifies the number of input signals that contribute to each output signal. Three methods based on the unitary Procrustes solution are proposed to enhance the spectral smoothness, the mix balance, and both properties jointly. A performance evaluation confirms the improvements of the mixing matrix in terms of objective measures. Furthermore, the evaluation results show that the error between the target and the generated coherence is lowered by increasing the spectral smoothness of the mixing matrix.
Grouped Feedback Delay Networks with Frequency-Dependent Coupling
(2023-05-17) Das, Orchisama; Schlecht, Sebastian J.; De Sena, Enzo
A1 Alkuperäisartikkeli tieteellisessä aikakauslehdessä
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.