Professional Documents
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Technical Report
NWU-EECS-09-08
May 11, 2009
Abstract
A simple but efficient stereo reverberation control is developed to simulate reverberation. Inspired from previous
classic studies, the reverberator is built using simple digital filters, including comb filters and all-pass filters commonly
used for digitally simulating reverberation. The digital reverberator incorporates low-pass filters to simulate the air and
walls absorption effect, gains to control the wet/dry effect and a delay difference parameter to introduce a difference
between the channels. A total of 21 digital parameters define the whole reverberation control.
Five measures commonly used to characterize reverberation are defined from the impulse response, and formulae are
derived to estimate values of these reverberation measures in terms of the digital parameters for the reverberator.
Dependence relations between the digital parameters lead to the need of only five independent parameters to control the
whole reverberator, so that the measures can be redefined as five functions of five independent parameters. This
mapping parameters-measures is used to let the reverberator to be controlled through the measures of the reverberation,
in other words the reverberator can generate a reverberation effect based on desired characteristics of the reverberation
itself.
Keywords: Reverberation, digital reverberator, comb filter, all-pass filter, Schroeder, Moorer,
reverberation time
A DIGITAL REVERBERATOR CONTROLLED
THROUGH MEASURES OF THE REVERBERATION
1. INTRODUCTION
In this section, three classic reverberators based on the digi- Similarly to Shroeder’s first reverberator, Chamberlin used
tal filters described in section 2 are presented and analyzed. five all-pass filters in series, with the last two filters doubled
Those digital reverberators are the first simple but efficient in parallel to get a stereo output in order to simulate a stereo
models of reverberation that have been proposed, and most of reverberation [12]. He also used un gain parameter to control
the work done on digital reverberation simulation is based on the overall gain of the reverberation, and finally summed the
those models. outputs of the stereo reverberation process to the stereo direct
sound.
3.1. Schroeder’s Reverberators The delays parameters should be approximately exponen-
tially distributed but in all cases must be a prime number of
By combining several of the digital filters described in sec- samples. As a first approximation, Chamberlin proposed to
tion 2, it is possible to achieve a more “natural” reverberation. give the first stage the longest delay, which is in the 50-msec
Schroeder first proposed to use five all-pass filters in series range, and then successively multiply it by a constant some-
with delay times that are incommensurate [10]. Each all-pass what less than 1. He precised that the last delay, the shortest
expands each impulse from the previous stage into an entire one, should not be much less than 10 msec if a distant, hollow
infinite all-pass impulse response. And as a series of all-pass sound is to be avoided. As for the gain values, they tend to be
filters is an all-pass filter, the whole unit still produces a “col- similar for all stages, although they should not be identical.
orless” reverberation. Respectively, delay times of 100, 68, Chamberlin explained that, for practical purposes, the rever-
beration time of a cascade of sections is equal to the longest
section time.
Finally, to simulate a stereo reverberation, a subtle differ-
ence was introduced between the parameters of each channel
to insure that the reverberation will be perceived as coming
from all directions, while the original signal retains its normal
directivity. Fig. 10 represents the scheme of Chamberlin’s re-
verberator.
controlling the block of parallel comb filters, in other words |Aa [n]| = ga da +1 for n = 0, da , 2da , ... (12)
12 control parameters. According to Moorer, the delay values Therefore, a single parameter control the all-pass filters at
are distributed linearly over a ratio of 1:1.5. The first comb each channel: the difference between the delay values m.
filter is defined as having the longest delay d1 , so the delays Note that to prevent exactly overlapping echoes, the delay
of the other comb filters can be deduced from d1 . Based on values for the comb filters and the all-pass filters are set to
previous studies (see section 3.3) and our own experiments, the closest inferior prime number of samples.
the range of values for the delays of the comb filters is de- For practical purposes, the low-pass filter following the
fined between 10 and 100 msec. This range is extended to the all-pass filter at each channel is defined from its cut-off fre-
values of d1 . Furthermore, according to Moorer, the rever- quency parameter fc [15]. fc has a non-inclusive range of val-
beration times (see section 5.1) of all the comb filters are set ues defined between 0 and half of the frequency sampling fs .
to be equal. Therefore, the gain values for the comb filters, Eq. 13 represents the gain gc of the low-pass filter (see sec-
can all be deduced from the reverberation time and the delay tion 2.3) computed from the corresponding cut-off frequency
value of the first comb filter. In particular, the gain factor of fc and frequency sampling fs .
the first comb filter g1 represents the smallest gain and has a s 2
non-inclusive range of values defined between 0 and 1. Al- fc fc
gc = 2 − cos 2π − cos 2π − 2 − 1 (13)
though the comb filter gives a non-flat frequency response, a fs fs
As for the gain parameter G following the low-pass filter at 5.1. The Reverberation Time
each channel, it is a simple temporal multiplicative constant
whose range of values is defined between 0 and 1. The Reverberation Time, T60 , is defined as the time in seconds
required for the reflections of a direct sound to decay by 60
In summary, a total of only five independent digital pa-
dB below the level of the direct sound [17].
rameters are needed to control the whole reverberation unit
and generate reverberation (The other parameters can be de- Note that the loudest crescendo for most orchestral music
duced from them according to the relations described above): is about 100 dB and a typical room background level for a
good music-making area is about 40 dB, thus the standard
1. The longest delay factor, of the first comb filter: d1 reverberation time can be seen as the time for the loudest
crescendo of the orchestra to decay to the level of the room
2. The smallest gain factor, of the first comb filter: g1 background [17].
The reverberation time for the comb filter and the all-pass
3. The delay difference between the all-pass filters at each filter defined in section 4 can be easily computed from their
channel: m discrete temporal function, respectively Eq. 11 and Eq. 12,
using the definition of the reverberation time above. Eq. 14
4. The cut-off frequency of the low-pass filters: fc and 15 represent respectively the reverberation time for the
comb filter and the all-pass filter. Note that, for both comb
5. The wet/dry gain parameter: G and all-pass filters, if the delay factor and the reverberation
time are given, the gain factor would be easy to deduce.
4.3. Other Improvements
log 10−3
T60k = dk (14)
Although the results are convincing reverberation effects, log gk
other improvements can still be made to produce an even
log 10−3
more realistic reverberation. More filters can be added to
T60a = da −1 (15)
achieve a greater echo density, or complex structures can be log ga
used to obtain a more “natural” sounding.
For example, Smith proposed the Digital Waveguide Net- The true value of the reverberation time of the reverbera-
works [16]. The idea is to build a network of bidirectional tion unit is measured from the impulse response of the rever-
delay lines simulating wave propagation in a duct capable of beration process, so without the direct sound at time t = 0.
producing the desired early reflections and a diffuse, suffi- Basically, the function looks for the last peak which has an
ciently dense late reverberation. Another example is the Feed- amplitude above 10−3 , which corresponds to −60 dB com-
back Delay Networks proposed by Jot and Chaigne [9]. They pared to the direct sound. Note that in discrete time, the direct
constitute a generalization of comb filters, where the number sound is represented as a Kronecker delta function of ampli-
of feedback paths have been increased using matrices. A last tude 1. The reverberation time would then be defined as the
example is the Late Reverberator of Gardner which uses the next sample to that detected peak, in seconds.
idea of nested all-pass filters [6]. To estimate the reverberation time directly from the pa-
Although, those improvements allow to achieve a more rameters of the reverberator, the impulse response of the
realistic reverberation, they also lead to more complex rever- whole reverberation unit needs to be seen as the sum of the
berators with multiple parameters, more complicated to ma- impulse responses of six parallel combinations of one comb
nipulate. filter, one all-pass filter, one low-pass filter and one wet/dry
gain filter in series, plus the direct sound which does not need
to be considered here since it is not included in the computa-
5. THE REVERBERATION MEASURES tion of the reverberation time. For each combination, every
peak of the comb impulse response is expanded into a whole
In this section, five measures commonly used to characterize all-pass impulse response. Note that since prime values of
reverberation, and defined from the impulse response of the delay have been used for the comb filters and the all-pass
reverberator are presented. Because those measures are more filters (see section 4), the echoes are assumed not to overlap
meaningful and less abstract representations of the reverbera- (or almost not). The effect of the next low-pass filter can be
tion, they are mapped to the control parameters of the rever- approximated as an additional gain of (1 − gc ) in the impulse
berator, so that the reverberator can eventually be controlled response corresponding to the amplitude gain of the low-pass
using the measures of the reverberation. Formulae are derived filter (see Eq. 8). The amplitude of the impulse response is
from the definitions of the measures in terms of the parame- further modified by a factor G corresponding to the wet/dry
ters of the digital filters, and the reverberation measures are gain. By making the approximation that the all-pass filter
eventually mapped to the five independent parameters defined simply adds an additional gain of ga in the impulse response
in section 4.2. corresponding to the gain factor of the all-pass filter, every
parallel combination can finally be seen as simple comb fil- events, but fuses them into a sensation of continuous decay
ter times an overall amplitude gain value of ga (1 − gc ) G. [18]. In other words, the early reflections become a late re-
Therefore, an approximation of the reverberation time for ev- verberation.
ery parallel combination can be computed from the definition
of the reverberation time of the comb filter in Eq. 14. 5.3. The Clarity
The reverberation time for the whole reverberation unit is
finally estimated as the maximum of the reverberation times The Clarity, Ct , describes the ratio in dB of the energies in the
of the parallel combinations, as shown in Eq. 16. impulse response p before and after a given time t [7]. The
definition of Ct in discrete time is given by the following Eq.
10−3
log g (1−g 19.
a c ) G t
T60 = max dk (16) X
p2 [n]
k=1..6 log gk
Ct = 10 log10 n=0
∞ (19)
X
2
5.2. The Echo Density p [n]
n=t
The Echo Density, Dt , is the frequency of occurring peaks at As its name suggests, the clarity can provide indication
a certain time t of the impulse response, in other words it is of how “clear” the sound is at time t. This time is usually
defined as the number of echoes per second at a time t [18]. referred to the arrival of the direct sound or the time when the
For one comb filter or one all-pass filter, the echo density early reflections give way to the late reverberation. t is usually
is independent from the time t and represents the delay value set to 50 msec for speech (C50 ) and 80 msec for music (C80 )
of the filter. Indeed sections 2.1 and 2.2 have shown that the [7].
delay factor determines the (constant) echo spacing in the im- For practical purposes, the clarity is here computed at time
pulse response of the filter. For parallel combinations of comb t = 0, corresponding to the arrival time of the direct sound.
or all-pass filters, the echo densities add, independently of t. In other words, the clarity represents now the ratio in dB be-
In the case of two comb and/or all-pass filters in series, the tween the energy of the direct sound and the energy of the
echoes overlap so that there is approximatively a linear echo reverberation. Since the direct sound is here represented as a
density buildup with time. Eq. 17 shows the estimation of Kronecker delta function of amplitude 1, the definition of the
the echo density for two filters in series with respective delay clarity becomes the minus of the energy of the reverberation
factors of dα and dβ . in dB, and the Eq. 19 is reduced to the following Eq. 20. Note
t that the linear clarity (not in dB) is called Definition and can
Dt = (17) be represented here simply as the inverse of the energy of the
dα dβ
impulse response.
By assuming that the low-pass filter does not introduce
any echo and since the wet/dry gain does not affect the echo ∞
X
spacing, the echo density of the whole reverberation unit can C = C0 = −10 log10 p2 [n] (20)
be seen as the echo density of six parallel combinations of n=0
one comb filter and on all-pass filter in series, not including The true value of the clarity of the reverberation unit is
the peak at t = 0 corresponding to the direct sound. There- measured from the impulse response of the reverberation pro-
fore, the echo density of the whole reverberation unit can be cess, so without the direct sound at time t = 0, by using the
estimated as follows, in Eq. 18. Note that the echo density Eq. 20 above.
is independent of any gain factor. For practical purposes, the To estimate the clarity directly from the parameters of the
echo density is computed at time t = 100 msec and defined reverberator, again the impulse response of the whole rever-
as D = Dt . beration unit needs to be seen as the sum of the impulse re-
6
t X 1 sponses of six parallel combinations of filters, plus the direct
D= (18) sound which does not need to be considered here since it is
da dk
k=1
not included in the computation of the clarity. Since prime
The true value of the echo density of the reverberation unit values of delay have been used for the comb filters and the
is measured from the impulse response of the reverberation all-pass filters so that the echoes do not overlap (or almost
process, so without the direct sound at time t = 0. Basically, not), it can be assumed that the sum of the squared amplitude
the function computes the number of peaks from the time 0 of the peaks of the impulse response of the whole reverbera-
till the time t, and divide this number by t to get the number tion process is equal to the sum of the squared amplitude of
of peaks per second. the peaks of the impulse responses of the parallel combina-
Note that if the echo density is larger than 20-30 echoes tions summed over all the six combinations. This assumption
per second, the ear no longer hears the echoes as separate is almost not altered by the low-pass filter effect. In other
words, the total energy of the impulse response of the six par- by Eq. 24 [7].
allel combinations can be considered to be equal to the sum
of the energies of the impulse responses of the combinations.
Furthermore, it can be shown that the total energy of the im-
pulse response of filters in series is equal to the product of the ∞
X
energies of the impulse responses of the filters. As a conse- np2 [n]
quence, the total energy of the impulse response for the whole n=0
TC = ∞ (24)
reverberation process can be approximated as a linear combi- X
2
p [n]
nation of the energies of the impulse responses of the filters n=0
which compose it.
Eq. 21 represents respectively the energies of the impulse
responses for one comb filter, the all-pass filter, the low-pass
filter and the wet/dry gain filter, all computed from their re-
spective discrete temporal functions from section 4. Note that
since the gain factor of the low-pass filter has been fixed to The true value of the central time of the reverberation unit
√1 , the energy of the impulse response of the low-pass filter is measured from the impulse response of the reverberation
2 process, so without the direct sound at time t = 0, by using
is equal to 1. Note also that the clarity for each of those filters
the Eq. 24 above.
can be easily computed from their respective energies of their
impulse responses as follows c = −10 log10 e.
To estimate the central time directly from the parameters
∞ ∞ of the reverberator, the impulse response of the whole rever-
X
2
X
2 i gk 2 beration unit needs to be seen as the sum of the impulse re-
e k = A k [n] = (gk ) =
0<gk <1 1 − gk 2
sponses of a series formed of the combination of six comb
n=0 i=1
∞ ∞ filters in parallel, the all-pass filter, the low-pass filter and the
ga 2
X X
Aa 2 [n] = (ga 2 )j+1 =
ea =
=1 wet/dry gain filter, plus the direct sound which does not need
0<ga <1 1 − ga 2
n=0 j=0 to be considered here since it is not included in the computa-
∞ ∞ tion of the central time. Since the echoes do not overlap, the
1 − gc
X X
ec = Ac 2 [n] = (1 − gc )2 (gc 2 )l =
same assumption made for the clarity is used here, namely
n=0
0<g c <1 1 + gc
l=0 “the square of the sum is equal to the sum of the squares”.
e G = G2
Based on this assumption and by using the definitions of the
(21) clarity in Eq. 23 and the discrete temporal function of the
Eq. 22 represents the total energy of the impulse response comb filter in Eq. 11, the central time of the combination of
of the whole reverberation process, in other words the six par- six parallel comb filters can be deduced as shown in Eq. 25.
allel combinations of one comb filter, the all-pass filter, the Note that for one single comb filter, the central time would be
dk
low-pass filter and the wet/dry gain filter in series. equal to tC k = 1−g 2.
k
6 6
X 1 − gc X gk 2
E= (ek ea ec eG ) = G2 (22)
1 + gc 1 − gk 2 ∞
6 X ∞
6 X 6
k=1 k=1 X X X dk gk 2
nAk 2 [n] idk (gk 2 )i
The clarity of the whole reverberation unit is finally esti- (1 − gk 2 )2
k=1 n=0 k=1 i=1 k=1
tC = 6 X∞
= 6 X∞
= 6
mated as the minus of the sum of the energies of the impulse X X X gk 2
responses of the six parallel combinations in dB, as shown Ak 2 [n] (gk 2 )i
1 − gk 2
in Eq. 23. Note that the clarity is independent of any delay k=1 n=0 k=1 i=1 k=1
factor. (25)
6
! Based on the same assumption, it can be shown that the
1 − gc X gk 2
2 central time of a series of filters is equal to the sum of the
C = −10 log10 G (23)
1 + gc 1 − gk 2 central times of the filters. By using the definitions of the
k=1
clarity and the discrete temporal functions of the all-pass filter
in Eq. 12 and the low-pass filter in Eq. 9, their respective
5.4. The Central Time central times can be deduced as shown in Eq. 26. Note that
since the gain factor of the low-pass filter has been fixed to
The Central Time, TC , is the “center of gravity” of the energy √1 , the central time of the low-pass filter is equal to da . Note
2
in the impulse response p, defined as follows in discrete time also that the central time is equal to 0 for the wet/dry gain
filter. the low-pass filter is first computed from its transfer function
∞ X∞ in Eq. 8, as shown in Eq. 29
jda (ga 2 )j+1
X
2
nAa [n] 1 − gc
da ga 2 Hc (n) = |Hc (ej2πn )| = p
n=0 j=0 (29)
t = = = = da 1 + gc 2 − 2 gc cos(2πn)
C
a ∞ ∞
1 − ga 2
X X
2 2 j+1
Aa [n] (ga )
The spectral centroid of the whole reverberation unit is
n=0 j=0
∞
finally estimated using the definition of the spectral centroid
∞
X X 1 of Eq. 28 and the magnitude frequency response of Eq. 29 as
nAc 2 [n] (1 − gc )2 (gc 2 )l
l
fs follows in Eq. 30. Note that gc is computed from fc so that
1 1
tC c = n=0 l=0
∞ = ∞ = the spectral centroid is assumed to only depend on the cut-off
X
2
X fs 1 − gc 2
(1 − gc )2 (gc 2 )l frequency parameter.
A c [n]
n=0 l=0
(26) fs /2
X n
The central time of the whole reverberation unit is finally 2 − 2 g cos(2πn)
estimated as the sum of the central times of the combination n=0
1 + gc c
SC = (30)
of six parallel comb filters, the all-pass filter and the low-pass fs /2
X 1
filter, as shown in Eq. 27. Note that the central time is inde- 2 − 2 g cos(2πn)
1 + gc c
pendent of any gain amplitude. Note also that the last term n=0
corresponding to the central time of the low-pass filter is very
small compared to the other terms and can be neglected. 5.6. Summary
6 Five measures that characterize the reverberation effect have
X dk gk 2
been defined in terms of all the parameters of the reverbera-
(1 − gk 2 )2 1 1 tion unit. Recall that a set of five digital parameters has been
k=1
TC = + da + (27)
X6
gk 2 fs 1 − gc 2 previously defined as controlling the whole reverberation unit.
1 − gk 2 The definitions of the five measures are then adapted accord-
k=1 ing to those five parameters.
The measures have been defined using the six delay and
5.5. The Spectral Centroid six gain parameters of the comb filters. In practice, the re-
lationships between the delays and the gains of the comb fil-
The Spectral Centroid, SC , is the center of gravity of the en- ters defined in section 4.2 reduce these numbers to the two
ergy in the magnitude spectrum P of the impulse response parameters of the first comb filter, d1 and g1 . Likewise, the
p, defined as follows in discrete time with the sampling fre- original fixed parameters of the all-pass filter, da and ga , have
quency fs by Eq. 28. Perceptually, it has a robust connection been used to define the measures. In practice, the measures
with the impression of “brightness” of a sound [19]. are computed for the left channel, where the delay parame-
fs /2
X ters of the all-pass filter are d7 = da + m2 and g7 = ga . The
nP2 [n] delay difference parameter m between the all-pass filters at
n=0 each channel is estimated from d7 . The parameters of the all-
SC = (28)
fs /2 pass filter for the right channel are then deduced as follows
d8 = da − m2 and g8 = ga . Note that the cut-off frequency
X
P2 [n]
n=0 fc and the wet/dry gain G are the same for each channel. Us-
ing Eq. 13, the cut-off frequency can be estimated from the
The true value of the spectral centroid of the reverberation
gain of the low-pass filter gc and the frequency sampling fs as
unit is measured from the magnitude of the impulse response
fs
of the reverberation process, so without the direct sound at follows fc = 2π arccos 2 − 12 (gc + g1 ) .
c
time t = 0, by using the Eq. 28 above. There are in summary five measures-functions of five
Although the comb filter has a non-flat magnitude fre- parameters-variables:
quency response, with a sufficient number of comb filters in
1. T60 (d1 , g1 , fc , G)
parallel, with equal values of reverberation time, an approx-
imated flat frequency response can be achieved. Since the 2. D(d1 , m)
all-pass filter has a flat magnitude frequency response, only
the low-pass filter is assumed to affect the frequency response 3. C(g1 , fc , G)
of the whole reverberation unit. 4. TC (d1 , g1 , m, fc )
To estimate the spectral centroid directly from the param-
eters of the reverberator, the magnitude frequency response of 5. SC (fc )
State-of-art mentions other common measures used to effect based on desired characteristics of the reverberation it-
characterize reverberation, such as the Energy Decay Time, self.
defined as the total signal energy remaining in the reverber- This work was supported by NSF grant number IIS-
ator impulse response p at time t, or the Inter-Aural Cross 0757544.
Correlation coefficient, which measures the difference in the
sounds arriving at the two channels. However, the Energy
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