Reverb.

qxd 3/7/07 11:53 AM Page 1

Reverberation
Reverberation
Reverberation
Reverberation and

Der everberation
Dereverberation
Dereverberation
Dereverberation
By Staff Technical Writer

I n this article we summarize recent AES convention papers dealing with reverberation and its artifi-
cial generation, analysis, and enhancement. How can some of the characteristics of reverberation be
measured in a perceptually relevant way? How can reverberation be removed successfully from
other wanted audio signals? It is also important to consider ways in which reverberation may be imple-
mented in platforms designed for interactive virtual environments, where computational resources may be
shared with visual processing and numerous changing sources have to be processed in real time.

ARTIFICIAL REVERBERATION frequency transform based on the short-

Although artificial reverberation pro-
cessors have been in existence for
many years, research is still continuing
into more efficient and better sounding
algorithms. Vickers et al., in “Fre-
quency Domain Artificial Reverbera-
tion Using Spectral Magnitude Decay”
(AES 121st paper 6926), explore the
concept of producing artificial rever-
beration in the frequency domain. Most
extant algorithms, they point out, are
either based on feedback-delay net-
works in the time domain or on convo-
lution in the frequency domain. The
former have a relatively low computa-
tional cost and provide control over
some perceptually relevant parameters;
but it is expensive to implement multi-
band equalizers in the feedback paths,
so any control over frequency-depen-
dent decay tends to be limited to a few
bands. On the other hand, those based
on convolution in the frequency
domain are very effective at simulating
specific physical spaces, but it is more
difficult to control individual parame-
ters, and such systems are computation-
ally expensive for long decay times. In
theory, the method described by the
authors would allow for detailed con-
trol over the frequency-dependent
decay time and require less memory
than feedback-delay networks.
The authors use a technique inspired
by the phase vocoder, shown in Fig. 1,
which essentially consists of a time-to-
time fourier transform (STFT),
followed by modifications to the phase x
and magnitude components of the windows
frequency spectrum, followed by an
inverse transform and reconstruction of
the time-domain signal. They investi-
fft(fftshift())
gate a range of techniques for generat-
ing reverberation and time-freezing
X
effects, mainly based on the accumula-
tion of successive frames of spectral
magnitude and phase information and Rectangular to Polar
the successive attenuation of the
magnitude components over time. They mag phase
find that the processing of phase infor-
mation in successive frames during the
Modifications
decay is crucial to the generation of a
perceptually natural reverberation. For
this reason they have to generate an
artificial phase signal that can be
combined with the the accumulated
Polar to Rectangular
magnitude response, such that the
reverb’s impulse response resembles
noise with an exponential decay. Some Y
of the issues to be overcome here fftshift(real(ifft))
include establishing the right tradeoff
between phase coherence and random-
ization, as well as the avoidance of
roughness and periodicity in the decay windows
structure. The authors find that an algo-
rithm with phase randomization applied
at the output works quite effectively
overlap-add
although the echo density does not
necessarily increase with time.
However, they argue that since late y
reverberation is normally said to begin Fig. 1. Phase vocoder (courtesy
where individual reflections cannot ➥ Vickers et al.)

J. Audio Eng. Soc., Vol. 55, No. 3, 2007 March 189

Reverb.qxd 3/7/07 11:53 AM Page 2

Reverberation and Der everberation

Fig. 3. Helical spring with driver and

receiver at either end

be heard, this does not seem to be a

problem. The quality of the output is
found to be reasonably good, though
not as high as that of the best reverbera-
tion devices based on feedback-delay
networks in the time domain or on
convolution in the frequency domain.
For this reason they plan further work
on aspects such as the control of modal
Fig. 2. Spring propagation modes: (a) transverse, (b) longitudinal, (c) torsional. density and the elimination of
(Figs. 2–6 courtesy Abel et al.) unwanted perceptual artifacts.
An alternative to the above types of
digital reverberation, stemming from
the early days of audio effects, is the
spring reverberator. Springs were origi-
nally used because they gave rise to
delays between the signal applied to an
exciter at one end and a receiver at the
other end of a spring. Abel et al., in
“Spring Reverb Emulation Using
Dispersive Allpass Filters in a
Waveguide Structure” (AES 121st
paper 6954), attempt to analyze and
emulate the performance of these clas-
sic devices using digital waveguide
models. They explain that springs are
approximately linear and time invariant
at typical operating levels for audio
systems, so they can be studied by
observing their impulse responses.
Springs can propagate waves in longitu-
dinal, transverse, or torsional modes (as
shown in Fig. 2). Modern devices typi-
Fig. 4. Typical spring reverb impulse response and spectrogram cally use the torsional mode with two or
three independent springs operating in
a parallel. Sometimes there are multiple
elements connected in series, or one
part of the spring is wound in the oppo-
site direction to the other, leading to
scattering at the junctions. A magnetic
driver at one end turns the spring so as
to set up a propagating torsional wave
b through the spring, which is detected by
a similar pickup at the other end, as
shown in Fig. 3. The impulse response,
shown in Fig. 4, tends to show a series
of decaying repetitions, but with each
Fig. 5. (a) Left- and right-going waves are processed separately using delay, reflected impulse having considerable
dispersion, and attenuation filters. (b) Separate sections of the spring reverb are and increasing smearing in the time
connected using scattering junctions. domain. This appears to be caused by
190 J. Audio Eng. Soc., Vol. 55, No. 3, 2007 March
Reverb.qxd 3/7/07 11:53 AM Page 3
Reverberation and Der everberation
frequency - KHz
frequency - KHz
Fig. 6. Comparison between measured (upper) and modeled (lower) impulse
response spectrograms for one particular spring reverb
Fig. 7. Overall system diagram for modified-CMA-based monaural dereverberation
(courtesy Huang and Kyriakakis)
low frequencies propagating faster than
high frequencies (see the spectrogram
in Fig. 4), which tends to turn impulses
into a chirp and eventually into a noise-
like sound. Little energy seems to prop-
agate above 4 kHz through most of the
springs that the authors tested.
The models used to emulate spring
reverbs, tried by the authors, had a struc-
ture based on that shown in Fig. 5,
consisting of a number of spring
sections connected using scattering junc-
tions. The results were reasonably
J. Audio Eng. Soc., Vol. 55, No. 3, 2007 March
successful, and good perceptual equiva-
lence is claimed between the models and
the original devices they aimed to
emulate. One example of the compari-
son between the spectrograms of
measured and modeled devices is shown
in Fig. 6, for the Accutronics Type 8.
DEREVERBERATION
Dereverberation is the identification and
removal of reverberation from other
wanted audio signals, using digital sig-
nal processing. Many different methods
have been tried. One of these is known
as blind dereverberation, in which the
algorithm only has access to the
received signal and has no knowledge
of the dry signal or acoustical environ-
ment giving rise to the reverberation.
Huang and Kyriakakis try a novel
approach based on blind deconvolution
in “Blind Dereverberation of Audio
Signals Using a Modified Constant
Modulus Algorithm” (AES 121st paper
6974). A constant modulus algorithm
(CMA) is used in conjunction with a
linear predictive coding (LPC) filter to
dereverberate a monophonic audio
signal. One of the problems that they
found in their previous work on this
topic was that typical constant modulus
algorithms assume that the input signal
is a statistically-independent and identi-
cally-distributed sequence, and that it
has a sub-Gaussian distribution with
negative kurtosis. (A Gaussian distribu-
tion is like the statistical distribution of a
white noise signal. Statistical distribu-
tion essentially describes the likelihood
that a signal will have a certain ampli-
tude. The degree of kurtosis describes
the peakedness of the distribution.)
Many audio signals, however, are quite
the opposite—they have non-white char-
acteristics and have positive kurtosis.
For this reason the authors employed
LPC analysis. Linear predictive coding
attempts to predict the current sample on
the basis of a weighted sum of previous
samples. The error or residual in this
process is the difference between the
prediction and the actual value of the
sample, which in this case is used in the
dereverberation process because the
LPC residual values tend to have a
“whiter” statistical characteristic than
the audio signal itself, with lower kurto-
sis. The residual signal is then used as
the input to the modified CMA algo-
rithm used for dereverberation. Because
of the linear nature of these algorithms
under the circumstances employed here,
the blind deconvolution filter thereby
derived can be applied directly to the
reverberant speech signal. This process
is shown in Fig. 7.
The authors found that this process
improved upon the results of a previ-
ous study in which they had used a
conventional CMA-based method, in
that the speech signal tested had less
reverberation, but they speculate ➥
191
Reverb.qxd 3/7/07 11:53 AM Page 4
Reverberation and Der everberation
that coloration and nonstationary prop-
erties of the speech signal might be
two problematic factors. They also
note that real room impulse responses
are nonminimum phase, which could
present difficulties with the type of
algorithm employed here.
In “A Perceptual Measure for
Assessing and Removing Reverb-
eration from Audio Signals” (AES
120th paper 6702), Zarouchas et al.
treat reverberation as a distortion or
degradation of an otherwise dry mono-
phonic audio signal. Unlike the method
of Huang and Kyriakakis, the process
they employ for dereverberation needs
access to the source signal and the
reverberant version of the same. A
computational auditory masking model
(CAMM) is employed to model the
perception of reverberant signals in a
similar way to sound quality measure-
ment approaches such as NMR (noise-
to-mask ratio) and PAQM (perceptual
audio quality measure) used for
measuring the perceptual significance
of codec distortions. As shown in Fig.
8, the source and reverberant versions
of the signal are both processed by the
CAMM, then a decision device makes
a subtraction of the “distortion” due to
reverberation on the basis of evaluat-
ing the just-noticeable difference
between the internal perceptual repre-
sentations of the two versions. In this
way only the perceptually-significant
reverberation is removed. The authors
report preliminary success with this
approach, which appears to be success-
ful at reducing the perceived reverber-
ance of a range of different material.
ANALYZING REVERBERATION
In two papers from the AES 121st
Convention (6928 and 6981), Bitzer
and Extra and colleagues present the
results of monaural and binaural anal-
ysis tools applied to artificial reverber-
ation. They attempted to find physical
metrics that could be used to predict
the perceived quality of artificial
reverberation devices, first of all using
traditional monaural measures of the
reverberation impulse response. The
metrics employed included the
energy-decay curve and the energy-
decay relief, the latter being based on
Jot’s 3-D surface plots of decay time
against frequency. In order to split the
192
Fig. 8. Dereverberation using a computational auditory model (CAMM) (Courtesy
Zarouchas et al.)
audio frequency range into bands, fil-
ters based on third octaves were used,
except that Bark-scaled bands were
employed below 500 Hz. These were
used to derive the reverberation time
and the early decay time in different
bands. Echo density was also mea-
sured as this has also been shown to
be an important factor governing the
perceived quality of reverberation.
The authors also examined short-time
histograms of the probability distribu-
tion of the decays to determine their
“whiteness” (see above), as this is
known to correspond to good-sound-
ing reverberation. The autocorrelation
function (the similarity of the signal to
itself at different time intervals), as
proposed by Griesinger, is a good way
of finding out whether there are repeti-
tive features in the reverberant decays,
as these tend to result in annoying
coloration. Objective clarity and time
variance were also measured.
The authors used these metrics on a
range of free and commercial reverbera-
tion devices and plug-ins and found that
they were able to observe the degree of
complexity in different algorithms.
They also conducted a listening test to
get some initial information about the
sound quality of different reverbs.
Initial results suggest that there is a rela-
tionship between the objective measures
and perceived quality, but these metrics
cannot currently replace a listening test.
A graininess and metallic quality was
observed in the sounds of some reverbs,
particularly on drum sounds, which
pointed the authors toward future devel-
opment ideas for metrics.
Examining binaural measures in a
simular way, Bitzer and Extra
attempted to find metrics that would
model some of the spatial characteris-
tics of artificial reverberation. IACC-
based measures, as well as lateral early
decay time (LEDT), room level (early)
(RLE), and a time-angle phase scope
were employed. The latter two were
found to be useful coarse predictors of
the quality of selected reverberation
algorithms. RLE is a measure that was
introduced by Trautmann and enhanced
by Blauert. It is based on an analysis of
short time segments of the first 80 ms
of the binaural impulse response. It
attempts to measure the energy in each
segment at a dominant angle of inci-
dence, compared with the frontal
energy. The time-angle phase scope is
based on the traditional Lissajous
figure display sometimes used to
analyze the coherence of stereo audio
channels, but in this case it is applied to
binaural impulse responses separated
into 40-ms overlapping blocks. The
result is a representation of the energy
and direction of the signal over time.
Abel and Huang provide further
insight into the development of metrics
for predicting reverberation quality in
“A Simple, Robust Measure of
Reverberation Echo Density” (AES
121st paper 6985). They work on a
similar premise to Bitzer and Extra and
colleagues, such that the late part of
reverberation has a smooth quality, a
Gaussian probability distribution, and a
high echo density. This is the result of a
suitably diffuse, mixed soundfield.
Echo density can be measured as a
function of time, they assert, by looking
at the standard deviation of the impulse
response in each time window, counting
the number of reflection “taps” that lie
outside this standard deviation, and
normalizing by that expected for
Gaussian noise. In the early part of the
decay where there are relatively few but
prominent reflections, the standard
deviation is large so a smaller number
of reflections are classified as outliers.
During the later, diffuse part of the
decay there will be a dense pattern of
overlapping reflections approximating
Gaussian noise. The authors found that
a time window of 20 to 30 ms worked
well as it is long enough to contain at
least a few reflections and short enough
J. Audio Eng. Soc., Vol. 55, No. 3, 2007 March
Reverb.qxd 3/7/07 11:53 AM Page 5
Reverberation and Der everberation
Fig. 9. Echo density profiles of feedback delay network impulse responses at
different diffusion settings: 0.2 (red), 0.4 (green), 0.6 (blue), and 1.0 (black), with
RT60 = 1.0s and roughly highpass equalization (courtesy Abel and Huang)
to have sufficient time resolution for
psychoacoustic purposes.
The resulting plots of echo density
according to the new measure were
found to discriminate well between
different diffusion settings of artificial
reverberators (see Fig. 9) and be robust
to different equalization and decay
parameters. It is claimed that since
traditional acoustical parameters for
measuring reverberation have not
included anything that accounts for the
temporal structure of reflections, this
new approach could be a useful indica-
tor of the time-domain quality or
texture of a reverberant signal.
REVERBERATION
ENHANCEMENT IN UPMIXING
Upmixing is the term commonly
applied to systems that attempt to
derive multichannel stereo signals
from two-channel or matrixed multi-
channel program material. Usher, in
“A New Upmixer for Enhancement of
Reverberance Imagery in Multichan-
nel Loudspeaker Audio Scenes” (AES
121st paper 6965), attempts to deal
with the problem that most upmixing
approaches are designed on the
assumption that stereo images are cre-
ated using amplitude-panned sources.
J. Audio Eng. Soc., Vol. 55, No. 3, 2007 March
Such upmixers tend to look for highly
correlated material between left and
right channels, assuming this repre-
sents front sources, extracting less cor-
related signals, and using them to
drive rear/side loudspeakers, assuming
they are diffuse sound or reverbera-
tion. However, as Usher points out,
some stereophonic recording tech-
niques use spaced microphones or
time-delay panning, and these record-
ings are not handled well by conven-
tional upmixers. He devised three
design criteria for a new upmixer: (1)
spatial distortion of the source image
in the upmixed audio scene should be
minimized; (2) reverberance imagery
Fig. 10. Conceptual diagram of Usher’s upmixer
should have a homogenous distribu-
tion in the horizontal plane, in particu-
lar, reverberance image directional
strength should be high from lateral
(±90) directions; (3) the new system
should not be dispreferred to a con-
ventional 2/0 system.
The system he developed uses an
adaptive filter and time-delay combi-
nation that is able to adjust input
signals in both frequency and time
before the difference signal between
the stereo channel pair is calculated
(see Fig. 10). By means of this
approach it was possible to upmix a
range of different stereo recordings to
four channel surround (using two addi-
tional loudspeakers at ±120°), such
that diagonally opposite loudspeakers
showed almost zero correlation but
side pairs had non-zero correlation. His
conjecture that this would maximize
the spatial fidelity of the front source
image while locating the reverberation
images to the sides was largely
supported by listening tests, and
upmixed results were generally
preferred to the original two-channel
versions.
REVERBERATION IN
ARTIFICIAL SCENE SYNTHESIS
Jot and Trivi take a novel approach to
the synthesis of reverberation for inter-
active virtual environments in “Scene
Description Model and Rendering
Engine for Interactive Virtual Acous-
tics” (AES 120th paper 6660). They
try to differentiate between the rever-
beration and reflection modeling that
might be required for interactive vir-
tual acoustics and that required for
applications such as architectural
acoustics. (The application in ques- ➥
193
Reverb.qxd 3/7/07 11:53 AM Page 6

Reverberation and Der everberation

tion is the Creative EAX environmen-
tal audio programming interface.) In
particular the authors are keen to
develop a means of generating reflec-
tions and reverberation that has mini-
T HE P RO C E EDINGth S mal computational costs, so they aim
to reduce the complexity of the algo-

OF THE A E S 29 rithms that might be used based on the

principle of plausibility. They argue,
for example, that when attempting to

IN T E R NATIONA L process large numbers of virtual sound

sources, an individual reflection rarely
has a critical effect on perception com-
C O N F ER E NC E pared to direct path components. They
therefore opt to prioritize the allocation
of resources toward improving the
shared reverberation process and the
control of per-source reverberator
feeds, only attempting to improve the
simulation of early or discrete reflec-
tions when necessary and if resources
allow.
In realistic virtual worlds, they
assert, audio accuracy is not as crucial
as in some other applications because
visual cues often dominate. Further-
more, virtual audio environments are
not attempting to simulate existing
situations so the requirement becomes
one of plausibility rather than accu-
racy. Auditory cues should be suffi-

A u d i o f o r M o b i l e a nd ciently valid and believable to support

H and he l d D e v i c e s
The 24 papers in this proceedings
address the important areas of
research and practical
applications—coding,
Class-D amplifiers, implementations,
speech processing, 3-D audio, and
synthetic audio—in the fast-paced,
emerging field of audio
for mobile and
handheld devices.
188 pages.
Also available on CD-ROM.
Purchase online at
www.aes.org/publications/conf.cfm
For more information email Donna Vivaro at dmv@aes.org
or call +1 212 661 8528, ext. 42.
the accompanying visual information.
The priority for resources therefore is
ordered like this: (1) direct path
components; (2) reverberation of
listener’s environment; (3) reverbera-
tion from other environments (such as
adjacent spaces); (4) refine early or
discrete reflections. For this reason a
reverberation system built on a physi-
cal model of the scene, with accurate
rendering of individual reflections, is
out of the question. The authors prefer
to contemplate traditional algorithms
based on feedback-delay networks or
possibly frequency-domain convolu-
tion with a measured or synthetic
reverb impulse response.
Editor’s note: The papers reviewed in this
article, and all AES papers, can be pur-
chased online at www.aes.org/publications/
preprints/search.cfm and www.aes.org/
journal/search.cfm. AES members also
have free access to a large number of past
technical review articles such as this one and
other tutorials from AES conventions and
conferences; go to www.aes.org/tutorials/.

194 J. Audio Eng. Soc., Vol. 55, No. 3, 2007 March