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FDNTB: The Feedback Delay Network Toolbox

Authors:

Abstract

Feedback delay networks (FDNs) are recursive filters, which are widely used for artificial reverberation and decorrelation. While there exists a vast literature on a wide variety of reverb topologies, this work aims to provide a unifying framework to design and analyze delay-based reverberators. To this end, we present the Feedback Delay Network Toolbox (FDNTB), a collection of the MAT-LAB functions and example scripts. The FDNTB includes various representations of FDNs and corresponding translation functions. Further, it provides a selection of special feedback matrices, topologies, and attenuation filters. In particular, more advanced algorithms such as modal decomposition, time-varying matrices, and filter feedback matrices are readily accessible. Furthermore, our toolbox contains several additional FDN designs. Providing MATLAB code under a GNU-GPL 3.0 license and including illustrative examples, we aim to foster research and education in the field of audio processing.
Proceedings of the 23rd International Conference on Digital Audio Effects (DAFx-20), Vienna, Austria, September 8–12, 2020
FDNTB: THE FEEDBACK DELAY NETWORK TOOLBOX
Sebastian J. Schlecht
Acoustics Lab, Dept. of Signal Processing and Acoustics
Media Lab, Dept. of Media
Aalto University, Espoo, Finland
sebastian.schlecht@aalto.fi
ABSTRACT
Feedback delay networks (FDNs) are recursive filters, which are
widely used for artificial reverberation and decorrelation. While
there exists a vast literature on a wide variety of reverb topologies,
this work aims to provide a unifying framework to design and an-
alyze delay-based reverberators. To this end, we present the Feed-
back Delay Network Toolbox (FDNTB), a collection of the MAT-
LAB functions and example scripts. The FDNTB includes vari-
ous representations of FDNs and corresponding translation func-
tions. Further, it provides a selection of special feedback matrices,
topologies, and attenuation filters. In particular, more advanced
algorithms such as modal decomposition, time-varying matrices,
and filter feedback matrices are readily accessible. Furthermore,
our toolbox contains several additional FDN designs. Providing
MATLAB code under a GNU-GPL 3.0 license and including il-
lustrative examples, we aim to foster research and education in the
field of audio processing.
1. INTRODUCTION
If a sound is emitted in a room, the sound waves travel through
space and are repeatedly reflected at the room boundaries resulting
in acoustic reverberation [1]. Many artificial reverberators have
been developed in recent years [2, 3], among which the feedback
delay network (FDN), initially proposed by Gerzon [4] and further
developed in [5, 6], is one of the most popular. The FDN consists
of Ndelay lines combined with attenuation filters, which are fed
back via a scalar feedback matrix A. Thus, any filter topology of
interconnected delays may be represented as an FDN in a delay
state space (DSS) representation, similar to the general state space
(SS) representations, which is an interconnection of unit delays.
Therefore, FDN framework provide the means for a systematic
investigation of a wide variety of filter topologies such as Moorer-
Schroeder [7], nested allpasses [8], allpass and delay combinations
[9], and many more. Equivalent structures are digital waveguides
[10], waveguide webs [11], scattering delay networks [12] and di-
rectional FDNs [13].
Artificial reverberation can be alternatively applied by directly
convolving the source signal with a room impulse response (RIR)
[2]. Whereas the general representation as a finite impulse re-
sponse (FIR) tends to imply higher computational costs, recent
developments yielded partitioned fast convolution schemes with
highly optimized implementation [14]. As any RIR, including the
FDN impulse responses, can be applied by convolution, it might
Copyright: © 2020 Sebastian J. Schlecht. This is an open-access article distributed
under the terms of the Creative Commons Attribution 3.0 Unported License, which
permits unrestricted use, distribution, and reproduction in any medium, provided the
original author and source are credited.
appear as a more general method. However, there are a few im-
portant advantages of FDNs. Where in principle, any combina-
tion of acoustic features can be applied by convolution as long as
they are linear and time-invariant, in practice numerical generation
or acoustic measurements are complex and involved topics [2].
Further, FDNs allow time-variation by modulating the input and
output gains and delays for early reflections of a moving source
[15, 12], and modulation of the feedback delays [16] and feed-
back matrix [17]. Also, synthesizing auditory scenes with mul-
tiple sources and multiple outputs for spatial reproduction scales
computationally well with FDNs compared to individual source-
to-receiver RIR convolution [18].
There are several central challenges in the design of FDNs,
which are only partly addressed in the research literature. A sig-
nificant challenge of FDN design is the inherent trade-off between
three aspects: computational complexity, mode density, and echo
density. Reduced modal density can lead to metallic sounding arti-
facts [19, 20], while reduced echo density can cause rough rattling
sounds. A higher number of delays increases both modal and echo
density, but also the computational complexity. Although attempts
have been made [21, 22], it remains open how to achieve spectrally
and temporally smooth FDNs with a minimal number of delays. A
closely connected topic is the choice of delay length. While the
co-prime criterium introduced by Schroeder [23] remains popular
and extensions exists [24], actual delay lines choices are still open
[25]. Recently, attempts have been made to quantify the spectral
quality of FDNs by statistical measures [26] and based on modal
decomposition [27], but perceptual verification and application ex-
amples need to be provided.
This work presents a Feedback Delay Network Toolbox (FD-
NTB) to support future research in this area. The toolbox contains
a wide variety of conversion functions between different FDN rep-
resentations such as delay state space, state space, modal, and ra-
tional transfer function. Further, matrix generation functions for
feedback matrices such as Hadamard, Circulant, and, random or-
thogonal are provided. Some well-known structures such as the
Moorer-Schroeder or nested allpasses are provided as well. Also,
the toolbox provides additional code for various example appli-
cations. Such applications include time-varying feedback matri-
ces, filter feedback matrices, proportional attenuation filters and
reverberation time estimation. We believe that supplying an entire
collection of different FDN approaches along with example appli-
cations within a unifying framework can be highly beneficial for
both researchers as well as educators in the field of audio process-
ing. The FDNTB can be found online1and is provided under the
GPL-3.0 license.
The remainder of this work is organized as follows. In Sec-
1https://github.com/SebastianJiroSchlecht/
fdnToolbox
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A(z)
x(z)b1(z)zm1c1(z)y(z)
b2(z)zm2c2(z)
b3(z)zm3c3(z)
d(z)
Figure 1: Filter feedback delay network (FFDN) with three delays, i.e., N3and single input and output, i.e., Nin 1and Nout 1,
respectively. All input, output and direct gains are potentially filters as well as the filter feedback matrix (FFM) Apzq.
tion 2, we review the general FDN structure, lossless and lossy
systems as well as topological variations. In Section 3, we review
a range of feedback matrix types and generation algorithms. Cor-
responding FDNTB function names are indicated as function.
2. FEEDBACK DELAY NETWORKS
In this section, we introduce FDN formally and review various
representations of FDNs. Further, we consider lossless and lossy
FDNs and the corresponding matrices and filters. This section is
concluded with an overview of topology variations of the standard
FDN.
2.1. Feedback Delay Networks (FDNs)
The single-input-single-output (SISO) FDN is given in time do-
main by the difference relation [28]
ypnq “ cJspnq ` d xpnq
spn`mq “ A spnq ` bxpnq,(1)
where xpnqand ypnqare the input and output values at time sam-
ple n, respectively, and ¨Jdenotes the transpose operation. The
FDN dimension Nis the number of delay lines and we occasion-
ally write N-FDN. The NˆNmatrix Ais the feedback matrix,
Nˆ1vector bof input gains, Nˆ1vector cof output gains
and scalar dis the direct signal gain. The lengths of the Nde-
lay lines in samples are given by the vector m“ rm1,...,mNs.
The Nˆ1vector spnqdenotes the delay-line outputs at time n.
The vector argument notation spn`mqabbreviates the vector
rs1pn`m1q,...,sNpn`mNqs. The system order of a standard
FDN in (1) is
N
N
ÿ
j1
mj.(2)
In Section 3, we discuss the time-varying feedback matrix Apnq
as effective manipulation of the resulting reverberation. Any gain
in Eq. (1) may consist of finite and infinite impulse response (FIR
and IIR) filters, for instance, a filter feedback matrix (FFM) Apzq
instead of a scalar feedback matrix A(see Fig. 1). The transfer
function of the filtered FDN (FFDN) in the z-domain, correspond-
ing to the difference relation in (1), is
Hpzq “ Ypzq
XpzqcpzqJrDm`z´1˘´Apzqs´1bpzq` dpzq,(3)
where Xpzqand Ypzqare the z-domain representations of the in-
put and output signals xpnqand ypnq, respectively, and Dmpzq “
diag`rz´m1, z´m2,...,z´mNs˘is the diagonal NˆNdelay
matrix. We abbreviate the loop transfer function with Ppzq “
Dm`z´1˘´Apzq. Alternatively, every gain and delay in (1) can
be time-varying to adjust to changing acoustic scenes.
2.2. Representations
There are multiple useful representations of FDNs. The represen-
tation (1) is called a delay state-space (DSS) representation and (3)
is the corresponding transfer function. Rocchesso [28] derived an
equivalent standard state-space (SS) representation, i.e., with all
delays equal to 1, see dss2ss. The matrix size of the equivalent
SS is then equal to Nin (2). The transfer function is equivalently
given by Hpzq “ qpzq
ppzq, where
qpzq “ dpzqdetpPpzqq ` cpzqJadjpPpzqqbpzq(4)
ppzq “ detpPpzqq,(5)
where adj denotes the matrix adjungate. The transfer function
form holds equally for the SS representation, see dss2tf and
dss2tfSym. Please note that in the multi-input-multi-output
(MIMO) case, ppzqis a scalar function, where qpzqis a ma-
trix which describes the input-output relation of size Nout ˆNin.
The polynomial coefficients are computed from the principal mi-
nors as given in [29, Lemma 1], see generalCharPoly and
generalCharPolySym.
The roots of the polynomial ppzqare the system poles,
whereas the roots of each entry of qpzqare the system zeros of
the corresponding input-output combination. The modal decom-
position of an FDN computes the partial fraction decomposition
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0 0.5 1 1.5 2 2.5 3
0
0.5
1
1.5
2
Mode Frequency [rad/sample]
Mode RT60 [s]
(a) Reverberation time of poles
0 0.5 1 1.5 2 2.5 3
120
110
100
90
80
70
Mode Frequency [rad/sample]
Mode Energy [dB]
(b) Magnitude of residues
Figure 2: Poles and residues for an 8-FDN with a frequency-dependent decay. Each dot indicates one mode in (a) frequency and reverber-
ation time and (b) frequency and residues magnitude. Delays are m“ r2300,499,1255,866,729,964,1363,1491sand Ais a random
orthogonal matrix.
of the transfer function in (3), i.e.,
Hpzq “
N
ÿ
i1
ρi
1´λiz´1,(6)
where ρiis the residue corresponding to the pole λi. Fig-
ure 2 shows the poles and residues of an FDN with frequency-
dependent reverberation time. The FDNTB provides the polyno-
mial Ehrlich-Aberth Method to compute the modal decomposi-
tion in (6), see dss2pr. Alternatively, the modal decomposition
can be computed from the SS or transfer function representation,
see dss2pr_direct. The residues in (6) can be computed in
two ways, directly from the transfer function [27, Section II-D],
dss2res or from the impulse response with a least-squares fit
[30], see impz2res.
Typically, the efficient DSS representation (1) is used to im-
plement the FDN, see dss2impz, but the impulse response can
be produced based on each of the representations. Therefore, we
provide also impulse responses from poles and residues as well
as the matrix transfer function representation, see pr2impz and
mtf2impz, respectively.
2.3. Lossless and Lossy Feedback
As a first step when designing, FDNs are commonly constructed
as lossless systems, i.e., all system poles lie on the unit circle [31].
The lossless property of general unitary-networks, which in par-
ticular applies to the FDN with a filter feedback matrix Apzq, was
described by Gerzon [31]. An FDN is lossless if Apzqis parauni-
tary, i.e., Apz-1qHApzq “ I, where Iis the identity matrix and ¨H
denotes the complex conjugate transpose [31]. For real scalar ma-
trices A, the FDN is lossless if Ais orthogonal, i.e., AJAI.
However also non-orthogonal feedback matrices may yield loss-
less FDNs [32, 33], and we give some examples in Section 3.
Homogeneous loss is introduced into a lossless FFDN by re-
placing each delay element z-1with a lossy delay filter γpzqz-1,
where γpzqis ideally zero-phase with a positive frequency-
response. The frequency-dependent gain-per-sample γpeıω qre-
lates to the resulting reverberation time T60pωqby
γpeıωq “ ´60
fsT60pωq,(7)
where fsis the sampling frequency and ωis the angular frequency
[5]. However, as substitution with lossy delays is impractical, the
attenuation filters are lumped into a single filter per delay line. In
standard FDNs with Apzq “ UΓpzq, where Uis lossless, the
delay-proportional attenuation filters should satisfy [5]
|Γpeıωq|diagpγpeıωqmqm(8)
where |¨|denotes the absolute value. There are various absorp-
tion filters ranging from the computationally efficient one-pole
shelving filter [5], see onePoleAbsorption, to highly accu-
rate graphical equalizers [34], see absorptionGEQ, and FIR
filters in absorptionFilters. With the modal decomposi-
tion (dss2pr), it is possible to demonstrate the lossless and lossy
behavior of FDNs. In Fig 2, a pole-residue decomposition is de-
picted for a 8-FDN with one-pole shelving filters designed with
T60 2s for the low and T60 0.4s for the high frequencies. Al-
though, the reverberation time follows the specified values, due to
the inaccurate magnitude and phase component, the reverberation
time can deviate from the specification.
2.4. Topology Variations
The absorption filters can be placed also directly after the delay
line such that the first pass through the delays are filtered as well.
In a similar manner, the feedback matrix is occasionally placed on
the forward path to increase the density on the first pass through.
Another related variation are extra allpass filters in the feedback
loop [35, 36]. Further, extra tap in and out points in the main delay
lines were proposed to increase the echo density and reduce the
initial time gap. More geometrically informed FDNs [12, 37, 13]
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may introduce extra filters and delays to account for source and
receiver positions.
Please note that any of the mentioned topology variations can
be represented and analyzed in formulation (3) by additional fil-
tering of the input and output gains. Although, the computational
complexity may differ from the optimal arrangements, in this work
we prioritize the comparability and generalizability and leave the
efficient implementation for the application scenario.
3. FEEDBACK MATRICES
In this section, we present a broad collection of feedback matrices
which are useful in the context of FDN design.
3.1. Lossless Feedback Matrices
The most important class of feedback matrices in FDNs are the
lossless matrices such that all system poles λiare on the unit cir-
cle. For the real matrices, we review special designs below, see
fdnMatrixGallery. To test statistically properties of FDNs,
we often choose random orthogonal matrices. Many random or-
thogonal matrix generators do not sample the space of possible
matrices equally, e.g., Gram-Schmidt orthogonalization. For this
reason, we provide the randomOrthogonal function, which
samples the space of all orthogonal matrices uniformly. The or-
thogonal matrices can be diagonally similar such that the lossless
property is retained (see diagonallyEquivalent) [32]. The
reverse process is less trivial, i.e., determining whether a matrix
Ais a diagonally similar to an orthogonal matrix. Such a pro-
cess might be necessary to determine whether a given matrix is
lossless. The provided algorithm isDiagonallySimilarTo-
Orthogonal is based on [41]. For instance, the allpass FDN
matrix (shown below) can be shown to be lossless, although not
orthogonal.
Alternatively, we can start with an arbitrary feedback matrix,
e.g., inspired by a physical design [12, 37], and find the nearest
orthogonal matrix. This so-called Procrustes problem solution is
provided in nearestOrthogonal. However, often the specifi-
cation does not necessary specify the phase (= sign) of the matrix
entries as it results from an energy-based derivation. For instance,
one might be interested in a feedback matrix which distributes
the energy from each delay line equally. As the conventional
Procrustes solution can give poor results, we have developed the
sign-agnostic version in [42] given nearestSignAgnostic-
Orthogonal.
Further, it is useful to interpolate between two given orthog-
onal matrices. However, the linear interpolation between matrix
entries does typically not yield orthogonal matrices. Instead, we
proposed to perform the interpolation for the matrix logarithms
[39]. The matrix exponentials map the antisymmetric matrices
to orthogonal matrices. Because the linear interpolation between
antisymmetric matrices remains antisymmetric, matrix exponen-
tial approach yields orthogonal interpolation matrices. We also
provide a special implementation of the inverse function, the ma-
trix logarithm, see realLogOfNormalMatrix based on [43].
For instance, interpolateOrthogonal allows to interpolate
between a Hadamard matrix and an identity matrix to adjust the
density of the matrix continuously and therefore the time-domain
density of the resulting impulse response [39].
3.2. Scalar Feedback Matrices
Here, we review a number of important scalar feedback matrices.
Table 1 lists many of the proposed matrices with the associated
operation counts. The implementation cost of the matrix-vector-
multiplication for a single time step vary from a conventional ma-
trix multiplication N2down to linear number of operations N.
Many of the examples are lossless matrices and loss is introduced
by additional attenuation filters. Some feedback matrices, most
notably from connections of allpass filters, do not have a lossless
prototype as the poles and zeros would cancel out at this limit case.
Some of the presented examples result from translating well-
known reverb topologies into a compact FDN representation.
Feedforward-feedback allpass filters have been introduced with the
delay lines to increase the short-term echo density [40, 7]. Alter-
natively, allpass filters may be placed after the delay lines [35, 36],
which in turn doubles the effective size of the FDN [29]. Gardner
proposed the nested allpass structure by [8], which recursively re-
places the delay in the allpass with another allpass. As a unified
representation, Fig. 4 depicts an overview of the present feedback
matrices.
3.3. Filter Feedback Matrices
If the sound is reflected at a flat, hard boundary, the reflection is co-
herent (specular), while it is incoherent (scattered) when reflected
by a rough surface. Towards a possible integration of scattering-
like effects in FDNs, we introduced in [33] the delay feedback ma-
trix (DFM), where each matrix entry is a scalar gain and a delay.
In [44], we generalized the feedback matrix of the FDN to a fil-
ter feedback matrix (FFM), which then results in a filter feedback
delay network (FFDN). As a special case of the FFM, we present
the velvet feedback matrix (VFM), which can create ultra-dense
impulse responses at a minimal computational cost [45].
FIR filter feedback matrices can be factorization as follows
Apzq “ DmKpzqUK¨ ¨ ¨ Dm1pzqU1Dm0pzq,(9)
where U1,...,UKare scalar NˆNunitary matrices and
m0,m1,...,mKare vectors of Ndelays. In this formulation,
the FFM mainly introduces Kdelay and mixing stages within
the main FDN loop. A few examples of FIR FFMs are de-
picted in Fig. 3. detPolynomial provides an efficient FFT-
based method for determining the polynomial matrix determinant
detpApzqq [46] for (5).
3.4. Time-Varying Feedback Matrix
Many FDN designs also introduce a time-varying component for
enhanced liveliness, improved time and modal domain density as
well as feedback stability in reverberation enhancement systems.
The most prominent variations are delay line modulation [16], all-
pass modulation [47] and matrix modulation [48, 39]. The all-
pass modulation can be represented equally as a matrix modulation
[39]. The matrix modulation is given by
Apn`1q “ ApnqR,(10)
where Ris an orthogonal matrix close to identity, see tiny-
RotationMatrix. A more robust and computationally efficient
version can be implemented by performing the modulation in the
eigenvalue domain, see timeVaryingMatrix.
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Name Definition Operations Counts Notes
Diagonal diagpvqNCorresponds to parallel comb filters
Orthogonal for |vi|1[38]
Triangular matrix
Lower (or Upper)
Lij 0for iăj NpN`1q{2or
N
May correspond to series comb filters
Lossless for |Lii|1
not orthogonal except diagonal [38]
Hadamard [6] H01
Hk`11
?2
»
HkHk
Hk´Hk
Nlog N
Fast Hadamard
transform
Orthogonal
|Hij |1
?N, i.e., equal magnitude
exists only for N1,2,4,8,12, . . .
Anderson [21] Sparse Block-Circulant matrix
KˆKblocks
NK Orthogonal, K4recommended
Sparse structures allows larger sizes
Householder [35] I´2vvJfor unit vector v2NOrthogonal
Symmetric
Circulant [28]
»
v1vN. . . v2
v2v1. . . v3
.
.
..
.
.....
.
.
vNvN´1. . . v1
|DFTpvq|1
2Nlog N`N
Fast Convolution
Orthogonal
Convolution is across channels not time
Random Orthogonal Uniform sampling of
orthogonal group OpNq
N2Orthogonal
useful for statistical tests
Tiny Rotation [39] AQ-1ΛQ
|=Λii|«ϵ
|Λii|1
N2Orthogonal
close to identity matrix Ifor small ϵ
Allows small matrix modifications
Diagonally Similar
Orthogonal [32]
D´1U D with
diagonal Dand orthogonal U
N2lossless, but not orthogonal
Allpasses in FDN
[35, 36]
Allpasses in
a FDN of size N{2
pN{2q2`NEquivalent to standard FDN of size N,
lossless, but not orthogonal
Moorer-Schroeder [7]
Reverberator
Series of N{2parallel comb and
N{2series allpasses
2NMoorer and Schroeder, Freeverb
Nested Allpass [8] Allpasses nested within allpasses NSISO allpass characteristic, not lossless
Table 1: Special matrices. The operations count are for a single matrix vector multiplication and are rough estimates as there are many
implementation details, e.g., the circulant matrix on a DFT implementation. Allpasses refer to Schroeder’s feedforward-feedback comb
filters [40, 36]. Some notation includes: |¨|denotes the absolute value; denotes a constant value for all parameters; and =denotes the
phase of a complex number.
4. CONCLUSIONS
In this paper, we have introduced the FDN toolbox (FDNTB), a
unifying MATLAB framework which contains several FDN al-
gorithms, various code examples for demo applications, as well
as additional measures that have already been used for evaluating
FDN algorithms. By doing so, we gave an overview on recent
studies and open topics. We hope that this toolbox not only pro-
vides a solid code basis to work in the field of FDN, but also helps
to direct attention to shortcomings of classical FDN algorithms, to
foster the development of new FDN techniques, and to ease the de-
sign of listening experiments. Finally, we would like to encourage
developers and researchers in the field of audio processing to use
the toolbox to realize their innovative FDN ideas.
5. ACKNOWLEDGMENT
The author thanks the anonymous reviewers for extensive and
helpful comments on the manuscript as well as the corresponding
software toolbox.
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Time [samples] - [1,17]
Amplitude [linear] - [-0.85,0.85]
(a) Delay feedback matrix (DFM) with K1.
Time [samples] - [1,413]
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(b) Velvet feedback matrix (VFM) with K2and δ
1
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... Unitary orthogonal matrices, such as the Hadamard or the Householder matrices [24], are typically chosen as the prototype for the feedback matrix A. In fact, being unilossless, they ensure stability regardless of the delays introduced in the FDN [25]. Then, losses are introduced by multiplying such a feedback matrix by a diagonal matrix containing scalar values designed to render a particular reverberation time [18]. ...
... However, as we will show in the next subsections, this does not prevent the method to minimize the loss function in (11) with optimal results. Finally, the baseline is implemented in MATLAB starting from the authors' codebase, which exploits the Feedback Delay Network Toolbox (FDNTB) [24], and we make use of MATLAB's Global Optimization Toolbox for finding m, B, C, and D. ...
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Recently, with the advent of new performing headsets and goggles, the demand for Virtual and Augmented Reality applications has experienced a steep increase. In order to coherently navigate the virtual rooms, the acoustics of the scene must be emulated in the most accurate and efficient way possible. Amongst others, Feedback Delay Networks (FDNs) have proved to be valuable tools for tackling such a task. In this article, we expand and adapt a method recently proposed for the data-driven optimization of single-input-single-output FDNs to the multiple-input-multiple-output (MIMO) case for addressing spatial/space-time processing applications. By testing our methodology on items taken from two different data-sets, we show that the parameters of MIMO FDNs can be jointly optimized to match some perceptual characteristics of given mul-tichannel room impulse responses, overcoming approaches available in the literature, and paving the way toward increasingly efficient and accurate real-time virtual room acoustics rendering.
... Notably, any orthogonal matrix is unilossless [23]. As such, Hadamard, Householder, and circulant matrices are widely used [19]. ...
... It is worth emphasizing that, although its parameters may indeed be frequency-independent, the FDN as a whole, belonging to a general class of recursive filters[19], is not. ...
Conference Paper
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Differentiable machine learning techniques have recently proved effective for finding the parameters of Feedback Delay Networks (FDNs) so that their output matches desired perceptual qualities of target room impulse responses. However, we show that existing methods tend to fail at modeling the frequency-dependent behavior of sound energy decay that characterizes real-world environments unless properly trained. In this paper, we introduce a novel perceptual loss function based on the mel-scale energy decay relief, which generalizes the well-known time-domain energy decay curve to multiple frequency bands. We also augment the prototype FDN by incorporating differentiable wideband attenua-tion and output filters, and train them via backpropagation along with the other model parameters. The proposed approach improves upon existing strategies for designing and training differentiable FDNs, making it more suitable for audio processing applications where realistic and controllable artificial reverberation is desirable, such as gaming, music production, and virtual reality.
... IV, the proposed method is applied to various FDN parameter sets and the SIMO case is presented. All figures have been made reproducible by including them in the Matlab FDN Toolbox, which is freely available [32]. ...
... Here, the inter-channel correlation of MIMO FDNs of three different sizes N = {4, 8, 16} is analyzed using six different feedback matrices. The Random Orthogonal, Hadamard, Householder, and Circulant matrix types are traditional frequency-independent gain matrices [32], whereas Velvet Scattering and Dense scattering are filter feedback matrices introduced recently [36]. The matrix types are defined in Appendix B. The computational costs of the Velvet scattering and Dense Scattering are equal, having K = 3 stages, with N K taps in each filter. ...
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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.
... Since the pioneering work of Schroeder and Logan [1], delaybased digital recursive structures have been used in reverberation synthesis [2]. Nowadays, one of the most widely used approaches in artificial reverberation is the feedback delay network (FDN), a system that generalizes the parallel comb-filter structure by interconnecting delays via a feedback matrix [3,4,5]. In FDNs, a commonly used approach is to first design a lossless prototype [6] to then achieve the desired frequency-dependent decay with attenuation filters [7,8]. ...
... The PyTorch implementation of the proposed method can be found in the dedicated repository 2 . A set of optimized FDN parameter values is readily available in the FDN Toolbox [5]. ...
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Artificial reverberation algorithms often suffer from spectral coloration, usually in the form of metallic ringing, which impairs the perceived quality of sound. This paper proposes a method to reduce the coloration in the feedback delay network (FDN), a popular artificial reverberation algorithm. An optimization framework is employed entailing a differentiable FDN to learn a set of parameters decreasing coloration. The optimization objective is to minimize the spectral loss to obtain a flat magnitude response, with an additional temporal loss term to control the sparseness of the impulse response. The objective evaluation of the method shows a favorable narrower distribution of modal excitation while retaining the impulse response density. The subjective evaluation demonstrates that the proposed method lowers perceptual coloration of late reverberation, and also shows that the suggested optimization improves sound quality for small FDN sizes. The method proposed in this work constitutes an improvement in the design of accurate and high-quality artificial reverberation, simultaneously offering computational savings.
... We used the MATLAB FDN toolbox [16] that provides an implementation of an FDN reverberation. This toolbox handles the synthesis of impulse response generation using delay state-space filter matrices, which reduce the processing time compared to sample-based implementation. ...
... From this desired frequency response H i ( f ) we then use the accurate equalizer developed by Valimaki et al. [8] using the least-squares method to get the corresponding cascaded second-order biquad filters coefficients. We used the implementation given in the FDN Toolbox [16] for that step. This process allowed us to have an initialization of the attenuation filter parameters that was already the best result. ...
Conference Paper
Full Text available here: http://www.aes.org/e-lib/browse.cfm?elib=21917 Recorded room impulse responses enable accurate and high-quality artificial reverberation. Used in combination with convolution, they can be computationally expensive and inflexible, providing little control to the user. On the other hand, reverberation algorithms are parametric which enable user control. However, they can lack realism and can be challenging to configure. To address these limitations, we introduce a multi-stage approach to optimize the coefficients of a Feedback Delay Network (FDN) reverberator to match a target room impulse response, thus enabling parametric control. In the first stage, we configure some FDN parameters by extracting features from the target impulse response. Then, we use a genetic algorithm to fit the remaining parameters to match the desired impulse response using a Mel-frequency cepstrum coefficients (MFCCs) cost function. We evaluate our approach across a dataset of impulse responses and conducted a subjective listening test. Our results indicate that the combination of the FDN with a short truncation of the target impulse response enables a better approximation, however, there are still differences with respect to the overall spectrum and the clarity factor in some more challenging cases.
... The idea of reverberation modelling with VN was continued by applying filtered sparse sequences to approximate segments of the late part of an RIR, at the same time closely following the target decay of each of such fragments [222,223]. Recently, VN signals were inserted in an FDN architecture to enhance the diffuseness of produced sounds [224,225,226]. ...
... The primary criterion for choosing the feedback matrix for an FDN is ensuring losslessness of the structure, i.e., the energy of the system should not decay when the attenuation filters are not in use [202,203,250]. Apart from this, the matrices are also used to enhance specific properties of the FDN, such as the increase in the echo density, computational efficiency, and spectral flatness [26,225,226,251,252,253]. ...
Thesis
Full-text available
In this dissertation, the discussion is centered around the sound energy decay in enclosed spaces. The work starts with the methods to predict the reverberation parameters, followed by the room impulse response measurement procedures, and ends with an analysis of techniques to digitally reproduce the sound decay. The research on the reverberation in physical spaces was initiated when the first formula to calculate room's reverberation time emerged. Since then, finding an accurate and reliable method to predict reverberation has been an important area of acoustic research. This thesis presents a comprehensive comparison of the most commonly used reverberation time formulas, describes their applicability in various scenarios, and discusses their accuracy when compared to results of measurements. The common sources of uncertainty in reverberation time calculations, such as bias introduced by air absorption and error in sound absorption coefficient, are analyzed as well. The thesis shows that decreasing such uncertainties leads to a good prediction accuracy of Sabine and Eyring equations in diverse conditions regarding sound absorption distribution. The measurement of the sound energy decay plays a crucial part in understanding the propagation of sound in physical spaces. Nowadays, numerous techniques to capture room impulse responses are available, each having its advantages and drawbacks. In this dissertation, the majority of commonly used measurement techniques are listed, whereas the exponential swept-sine is described in more detail. This work elaborates on the external factors that may impair the measurements and introduce error to their results, such as stationary and non-stationary noise, as well as time variance. The dissertation introduces Rule of Two, a method of detecting nonstationary disturbances in sweep measurements. It also shows the importance of using median as a robust estimator in non-stationary noise detection. Artificial reverberation is a popular sound effect, used to synthesize sound energy decay for the purpose of audio production. This dissertation offers an insight into artificial reverberation algorithms based on recursive structures. The filter design proposed in this work offers precise control over the decay rate while being efficient enough for real-time implementation. The thesis discusses the role of the delay lines and feedback matrix in achieving high echo density in feedback delay networks. It also shows that four velvet-noise sequences are sufficient to obtain smooth output in interleaved velvet noise reverberator. The thesis shows that the accuracy of reproduction increases the perceptual similarity between measured and synthesised impulse responses. The insights collected in this dissertation offer insights into the intricacies of reverberation prediction, measurement and synthesis. The results allow for reliable estimation of parameters related to sound energy decay, and offer an improvement in the field of artificial reverberation.
... All code to reproduce the figures of this letter can be found online [34]. The core functions are included in the FDN Toolbox [35]. ...
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Feedback delay networks (FDNs) are used in audio processing and synthesis. The modal shapes of the system describe the modal excitation by input and output signals. Previously, the Ehrlich-Aberth method was used to find modes in large FDNs. Here, the method is extended to the corresponding eigenvectors indicating the modal shape. In particular, the computational complexity of the proposed analysis method does not depend on the delay-line lengths and is thus suitable for large FDNs, such as artificial reverberators. We show the relation between the compact generalized eigenvectors in the delay state space and the spatially extended modal shapes in the state space. We illustrate this method with an example FDN in which the suggested modal excitation control does not increase the computational cost. The modal shapes can help optimize input and output gains. This letter teaches how selecting the input and output points along the delay lines of an FDN adjusts the spectral shape of the system output.
... All code to reproduce the figures of this letter can be found online 1 . The core functions are included in the FDN Toolbox [34]. ...
Preprint
Full-text available
Feedback delay networks (FDNs) are used in audio processing and synthesis. The modal shapes of the system describe the modal excitation by input and output signals. Previously, the Ehrlich-Aberth method was used to find modes in large FDNs. Here, the method is extended to the corresponding eigenvectors indicating the modal shape. In particular, the computational complexity of the proposed analysis method does not depend on the delay-line lengths and is thus suitable for large FDNs, such as artificial reverberators. We show the relation between the compact generalized eigenvectors in the delay state space and the spatially extended modal shapes in the state space. We illustrate this method with an example FDN in which the suggested modal excitation control does not increase the computational cost. The modal shapes can help optimize input and output gains. This letter teaches how selecting the input and output points along the delay lines of an FDN adjusts the spectral shape of the system output.
... This method can be seen as a frequency-dependent extension of the former median and dc gain factors. This letter shows that the proposed two-stage attenuation filter improves the accuracy of approximation in feedback-based reverberators, such as the feedback delay network (FDN) [2], [19], [20], [21], the scattering delay network [22], [23], and the interleaved velvet-noise (IVN) reverberator [4]. The proposed method is easy to apply automatically, as the pre-filter does not require optimization, but simply adjusting the gain at two frequency points. ...
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Full-text available
Delay networks are a common parametric method to synthesize the late part of the room reverberation. A delay network consists of several feedback loops, each containing a delay line and an attenuation filter, which approximates the same decay rate by appropriately setting the frequency-dependent loop gain. A remaining challenge is the design of the attenuation filters on a wide frequency range based on a measured room impulse response. This letter proposes a novel two-stage attenuation filter structure, sharpening the design. The first stage is a low-order pre-filter approximating the overall shape and determining the decay at the two ends of the frequency range, namely at the dc and the Nyquist limit. The second filter, an equalizer, fine-tunes the gain at different frequencies, such as on one-third-octave bands. It is shown that the proposed design is more accurate and robust than previous methods. A design example applying the proposed method to an interleaved velvet-noise reverberator is also exhibited. The proposed two-stage attenuation filter is a step toward a realistic parametric simulation of measured room impulse responses.
... The two-stage decay plots of the coupled RIRs as the aperture size increases are shown in Fig. 5. All RIRs have been generated with the FDNToolbox [36]. As the aperture increases, the early decay generally becomes shorter for the RIRs. ...
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Full-text available
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 paper, 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.
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The mixing matrix of a Feedback Delay Network (FDN) reverberator is used to control the mixing time and echo density profile. In this work, we investigate the effect of the mixing matrix on the modes (poles) of the FDN with the goal of using this information to better design the various FDN parameters. We find the modal decomposition of delay network reverberators using a state space formulation, showing how modes of the system can be extracted by eigenvalue decomposition of the state transition matrix. These modes, and subsequently the FDN parameters, can be designed to mimic the modes in an actual room. We introduce a parameterized orthonormal mixing matrix which can be continuously varied from identity to Hadamard. We also study how continuously varying diffusion in the mixing matrix affects the damping and frequency of these modes. We observe that modes approach each other in damping and then deflect in frequency as the mixing matrix changes from identity to Hadamard. We also quantify the perceptual effect of increasing mixing by calculating the normalized echo density (NED) of the FDN impulse responses over time.
Conference Paper
Full-text available
This paper received the best paper award at WASPAA 2019. Feedback delay networks (FDNs) belong to a general class of re-cursive filters which are widely used in artificial reverberation and decorrelation applications. 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 grows polynomially over time, and that the growth depends on the number and lengths of the delays. In this work, we introduce so-called delay feedback matrices (DFMs) where each matrix entry is a scalar gain and a delay. While the computational complexity of DFMs is similar to a scalar-only feedback matrix, we show that the echo density grows significantly faster over time, however, at the cost of non-uniform modal decays.
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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.
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Artificial reverberation algorithms generally imitate the frequency-dependent decay of sound in a room quite inaccurately. Previous research suggests that a 5% error in the reverberation time (T60) can be audible. In this work, we propose to use an accurate graphic equalizer as the attenuation filter in a Feedback Delay Network re-verberator. We use a modified octave graphic equalizer with a cascade structure and insert a high-shelf filter to control the gain at the high end of the audio range. One such equalizer is placed at the end of each delay line of the Feedback Delay Network. The gains of the equalizer are optimized using a new weighting function that acknowledges nonlinear error propagation from filter magnitude response to reverberation time values. Our experiments show that in real-world cases, the target T60 curve can be reproduced in a perceptually accurate manner at standard octave center frequencies. However, for an extreme test case in which the T60 varies dramatically between neighboring octave bands, the error still exceeds the limit of the just noticeable difference but is smaller than that obtained with previous methods. This work leads to more realistic artificial reverberation.
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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.
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The most efficient binaural acoustic modeling systems use a multi-tap delay to generate accurately modeled early reflections, combined with a feedback delay network that produces generic late reverberation. We present a method of binaural acoustic simulation that uses one feedback delay network to simultaneously model both first-order reflections and late reverberation. The advantages are simplicity and efficiency. We compare the proposed method against the existing method of modeling binaural early reflections using a multi-tap delay line. Measurements of ISO standard evaluators including interaural correlation coefficient, decay time, clarity, definition, and center time, indicate that the proposed method achieves comparable level of accuracy as less-efficient existing methods. This method is implemented as an iOS application, and is able to auralize input signal directly without convolution and update in real time.
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In many applications, it is desirable to achieve a signal that is as close as possible to ideal white noise. One example is in the design of an artificial reverberator, whereby there is a need for its lossless prototype output from an impulse input to be perceptually white as much as possible. The Ljung-Box test, the Drouiche test, and the Wiener Entropy—also called the Spectral Flatness Measure—are three well-known methods for quantifying the similarity of a given signal to ideal white noise. In this paper, listening tests are conducted to measure the Just Noticeable Difference (JND) on the perception of white noise, which is the JND between ideal Gaussian white noise and noise with a specified deviation from the flat spectrum. This paper reports the JND values using one of these measures of whiteness, which is the Ljung-Box test. This paper finds considerable disagreement between the Ljung-Box test and the other two methods and shows that none of the methods is a significantly better predictor of listeners' perception of whiteness. This suggests a need for a whiteness test that is more closely correlated to human perception.
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Discrete-time rational transfer functions are often converted to parallel second-order sections due to better numerical performance compared to direct form infinite impulse response (IIR) implementations. This is usually done by performing partial fraction expansion over the original transfer function. When the order of the numerator polynomial is greater or equal to that of the denominator, polynomial long division is applied before partial fraction expansion resulting in a parallel finite impulse response (FIR) path.
Thesis
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In today's audio production and reproduction as well as in music performance practices it has become common practice to alter reverberation artificially through electronics or electro-acoustics. For music productions, radio plays, and movie soundtracks, the sound is often captured in small studio spaces with little to no reverberation to save real estate and to ensure a controlled environment such that the artistically intended spatial impression can be added during post-production. Spatial sound reproduction systems require flexible adjustment of artificial reverberation to the diffuse sound portion to help the reconstruction of the spatial impression. Many modern performance spaces are multi-purpose, and the reverberation needs to be adjustable to the desired performance style. Employing electro-acoustic feedback, also known as Reverberation Enhancement Systems (RESs), it is possible to extend the physical to the desired reverberation. These examples demonstrate a wide range of applications where reverberation is created and enhanced artificially employing signal processing techniques. A major challenge of designing artificial reverberators is the high complexity of the physical reverberation process. Even small office spaces of 40 m^3 exhibit more than 10^7 acoustic modes, in concert halls the number of acoustic modes can surpass 10^9 in the audible range. The room geometry, as well as the interaction with the boundary materials, can be as well fairly complex. Whereas these complex considerations are mandatory for simulations of specific spaces, used for example for the acoustic and architectural planning of a concert venue, they are somewhat misleading in the realm of artistic applications. The focus on perceptually convincing artificial reverberation algorithms provides the freedom to make some simplifications to the generation process, leading to the recursive systems, which play a central role in this dissertation. Two specific formulations of recursive systems for artificial reverberation are considered: Firstly, Feedback Delay Networks (FDNs) which are built around multiple delays which are fed back to their inputs and by this mimic the recursive process of sound waves bouncing back and forth in an acoustic space. And secondly, RESs, which are installed in rooms to extend the physical reverberation via electro-acoustic feedback between microphones and loudspeakers. The main objective of artificial reverberators is to recreate and enhance room impulse responses while considering three aspects: i) accurate recreation of physical spaces; ii) delivering perceptually convincing spaces; and iii) efficiency of processing and parameterization. The primary goal of this dissertation is to achieve better control over the evolution of the artificial reverberation over time, namely the evolution of normal modes and reflections over time. The decay rate of normal modes most importantly determines the stability of the system, but also the perceptual quality of the artificial reverberation. For this purpose, existing network topologies for artificial reverberation are unified in the general FDN framework. For the FDN, an analytic formulation of the polynomial governing the recursive behavior is presented from which analytic constraints on the angular distribution of the decaying modes are derived. Lossless FDNs are commonly used as a design prototype for artificial reverberation algorithms for which all normal modes neither decay nor rise. The lossless property is dependent on the feedback matrix, which connects the output of a set of delays to their inputs, and the lengths of the delays. This work presents the most general class of feedback matrices which constitutes lossless FDNs regardless the lengths of the delays. As a secondary goal, the temporal features of impulse responses produced by FDNs, i.e., the number of echoes per time interval and its evolution over time, are analyzed. This so-called echo density is related to known measures of mixing time and their psychoacoustic correlates such as perception of the room size. It is shown that the echo density of FDNs follows a polynomial function, whereby the polynomial coefficients can be derived from the lengths of the delays for which an explicit method is given. The mixing time of impulse responses can be predicted from the echo density, and conversely, the desired mixing time can be achieved by a derived mean delay length. In the last part of this dissertation, a novel time-variant reverberation algorithm is introduced. By modulating the feedback matrix nearly continuously over time, an intricate pattern of concurrent amplitude modulations of the feedback paths evolves. It is demonstrated that the perceived quality of the decaying normal modes can be enhanced by the feedback matrix modulation. The same technique of time-varying feedback matrices is applied in multichannel sound systems to improve the system's stability. It is shown with a statistical approach that time-varying mixing matrices can achieve optimal stability improvement for a higher number of channels. A listening test demonstrates the improved quality of time-varying mixing matrices over comparable existing techniques.