Calculating Reverberation Time From Impulse Responses - A Comparison of Software Implementations
Calculating Reverberation Time From Impulse Responses - A Comparison of Software Implementations
DOI 10.1007/s40857-016-0055-6
TECHNICAL NOTE
Received: 1 February 2016 / Accepted: 16 April 2016 / Published online: 4 May 2016
© Australian Acoustical Society 2016
Abstract In room acoustics measurement, calculating reverberation time from room impulse responses is often done, aided
by software. This paper compares the performance of nine software implementations for calculating octave band reverberation
time, including two written by the authors. Synthetic impulse responses are used to test decays without and with a steady
noise floor, and an impulse response from a real measurement is also used for comparison. Results indicate no significant
reverberation time calculation problems for noise-free exponential decays, and for exponential decays leading to a steady noise
floor. Frequency selectivity is identified as an area for potential improvement in filter-bank design, and a highly selective octave
band filter-bank is shown to be effective without introducing errors. Testing with a real measured impulse response, which
had been used in a 2004 study comparing reverberation time analysis implementations, showed greater agreement between
software than was found previously. This might reflect an improvement in software performance in the years between the
two studies. However, it also might reflect the smaller scale of the present study. Nevertheless, the results can contribute to
confidence in current software implementations.
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Lundeby et al. [7] and Bradley [8] conducted studies 1.1 Issues in the Calculation of Reverberation Time
in the 1990s comparing systems for reverberation time
measurement and analysis. In Lundeby’s study the whole In general, the procedure for calculating reverberation time
measurement and analysis system (including physical equip- from a single measured room impulse response will involve
ment) was included, whereas Bradley used an electronic the following steps:
artificial reverberator to remove physical variability. How-
ever, both studies were comparisons of measurement and 1. Truncation of the impulse response at the start, although
analysis systems, rather than isolating just the analysis part. this should have little effect on reverberation time calcu-
Subsequent studies by Ikegami and Uchida [9] and Katz lation and is mainly important for energy ratio parameters
[10] focussed only on analysis by using an impulse response such as clarity index (ISO3382-1 suggests starting where
as the test input for software. Katz’s impulse response (a the energy first rises significantly above the background
recording of a balloon pop in a large room) is of partic- noise, but is more than 20 dB below the peak);
ular interest because of the large scale of the study and 2. Applying a filter-bank (e.g. octave band);
because the impulse response has a significant noise floor. 3. Truncation of the impulse response at the end, usually at
The impulse response was sent to researchers and practition- a different point in each band;
ers to analyse, using software of their choice. Results showed 4. Squaring the impulse response;
a wide range of reverberation time values in the lowest octave 5. Optionally compensating for the decay energy lost from
band (125 Hz), for which signal-to-noise ratio was poorest. end-truncation and/or subtraction of the excess energy
Katz’s study is also of particular interest because some of due to a steady-state noise floor;
the software that was used for the tests is still in use today, 6. Integrating the impulse response in reverse time, and con-
albeit in updated versions. Since Katz’s study, some smaller verting to decibels; and
scale software comparison studies have been published, for 7. Finding reverberation time parameters by linear regres-
example, by Tronchin et al. [11] (four software implemen- sion over the appropriate evaluation ranges (−5 to
tations), Ţopa et al. [12] (three implementations), Machín −25 dB for T20 ; −5 to −35 dB for T30 ).
et al. [13] (three implementations, two external and one by
the authors) and Mansilla et al. [14] (four implementations, Steps 2 and 3 and the optional step 5 have some scope
three external and one by the authors). The conclusions of for variations in approach, which could affect the resulting
these studies vary: Tronchin et al., using impulse responses values.
measured in an opera theatre, found average T30 (reverbera- Filter-banks for reverberation time measurement can
tion time) deviations of up to 8.3 % between software, with a influence the result for two reasons. Firstly, the time-response
maximum deviation of almost 25 %; Ţopa et al. tested three of the filter-bank can have its own decay, which could artifi-
rooms (a school hall, a church and a university auditorium) cially increase the measured reverberation time [16]. This is
and found that the software did not affect the results; the only likely to be noticeable when reverberation time is very
main aim of the study by Machín et al. was validate soft- short, and especially in the lower octave or 1/3-octave bands.
ware developed by the authors through a set of measurements However, this can be avoided by using reverse-time filter-
in one auditorium, and they found only minor deviations in ing [3,17]. Secondly, the selectivity of the filter-bank is only
reverberation time between the tested software; and Mansilla loosely defined by standards [18–20], and cases where there
et al. tested 59 diverse impulse responses, finding average is significant variation in reverberation as a function of fre-
octave band deviations of about 0.3 s in T30 between the quency will be better represented by the results of a highly
external software tested. Concurrent with the present study, selective filter-bank. Venturi et al. [21] suggested the use
Álvarez-Morales et al. [15] conducted a detailed software of highly selective filters (144 dB/octave skirts, equivalent to
comparison based on a large set of in-situ measurements 24th order) for octave band reverberation time measurement,
in a university auditorium, using four software implemen- finding that high-order filtering always improved the accu-
tations, with extensive statistical analysis. They found that racy reverberation time results compared to sixth-order filters
the implementations mostly agreed when used as recom- (both of which comply with IEC Class 0 response limits from
mended by the software documentation. Hence the extent the standard at the time [19]). However, they speculated that
to which software agrees or disagrees appears to depend on lower selectivity might be needed for very short reverbera-
the type of testing done, and the present paper focuses on test- tion times and for short evaluation ranges such as early decay
ing specific potential vulnerabilities of software, along with time and T10 .
a reassessment using Katz’s impulse response with present Much has been written about end-truncation, because it
day software. has a large effect on the results in practical measurements
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[4,6,7,22]. If end-truncation is omitted, then the background value due to lack of validity. Where available, noise com-
noise will accumulate in the reverse integration, potentially pensation is applied (including in cases where the impulse
yielding artificially high reverberation time values. Follow- response is essentially noise-free), and where reverse-time
ing ISO3382-1, the end-truncation point should be at the filtering is chosen if it is a user setting. For the sake of
intersection between a horizontal line representing the back- succinctness, only T30 values are reported (i.e. reverbera-
ground noise and a sloping line representing the level decay tion time calculated over the −5 to −35 dB evaluation range
(prior to reverse integration) [18], but the method for finding in the reverse-integrated decay), as these are more vulner-
this could vary between implementations, and it is likely that able than T20 to background noise, truncation and filter
variation in results between software for impulse responses response. Indeed, Hak et al. [5] show that T30 is the most
that include noise will be partly due to the way in which sensitive of the ISO 3382-1 parameters to signal-to-noise
end-truncation is done. ratio.
Compensation for the energy lost from end-truncation is Seven software implementations were tested, together
an optional step in ISO3382-1, which helps straighten the with two of the authors’ own implementations. The seven
reverse-integrated decay. Without doing this, there may be external software instances in this study include prominent
an underestimation of reverberation time. The specifics of reverberation time calculation software that tends to be used
how this is done are open to interpretation, but ISO3382-1 in professional and research work: in alphabetic order, the
recommends a value derived from the decay rate in the final software is ARTA, Aurora, Dirac, EASERA, Odeon, Smaart
10 dB. A variation of this, by Lundeby et al. [7], provides and WinMLS. Software that required a superseded operat-
a 5–10 dB safety margin in finding this late decay rate, to ing system or particular hardware was excluded. All of this
reduce the influence of the noise floor on the late decay rate software has been included in previous software compar-
calculation. isons by: Katz (Aurora, Dirac, Smaart and WinMLS) [10],
Noise power subtraction is another technique that helps Tronchin et al. and Ţopa et al. (Aurora, Dirac and Win-
straighten the reverse-integrated decay [2]. Although this MLS) [11,12], Machín (Odeon, WinMLS) [13], Mansilla
approach is not compliant with ISO3382, noise compensation et al. (Aurora, Dirac, EASERA) [14] and Álvarez-Morales
is implemented (although not necessarily described) in com- (ARTA, Dirac, EASERA, WinMLS) [15]. One of the con-
monly used software. Guski and Vorländer [6] recommended cerns with software selection was that the software be
that this, together with truncation energy compensation, recognised as a serious analysis tool, and inclusion in pre-
be included in future modification of ISO3382. Where vious software comparisons done by acoustics researchers
available, it was included in the software tests reported provides one indication of this. Two other software imple-
here. mentations, which were not in these previous comparison
There are many non-standard alternative approaches to studies, were tested in preparing the present paper, but were
calculate reverberation time from an impulse response. excluded from this study due to the combination of deficien-
For example, Xiang [23] proposed non-linear regression cies in performance (yielding outlying values in the tests)
accounting for the reverse-integrated noise floor. Morgan and a lack of external evidence that the software was recog-
[4] proposed using an evaluation range that depended upon nised as a serious analysis tool (such as multiple citations
the noise floor (finishing 5 dB above the noise, in conjunc- in room acoustics literature or authorship associated with
tion with end-truncation at the elbow) instead of using fixed acoustics research). Although Smaart tends to be used more
evaluation ranges. However, mostly reverberation calcula- for professional audio system tuning than for room acoustics,
tion software aims to follow a standard approach (often it was included in the study because of its inclusion in
with the non-standard option of noise power subtraction), Katz’s. Every software version tested was current in Decem-
and it is common to see references to ISO3382 in such ber 2015. Like some previous software comparison studies
software. in room acoustics, including Katz’s, the external software is
de-identified, and is referred to arbitrarily using the letters
A to G.
2 Software and Test Concept The two implementations that were developed by the
authors and their colleagues are the ITA Toolbox (developed
The test concept of the present study is that the software in the Institute of Technical Acoustics, RWTH University
settings are fixed (the same settings used for every impulse Aachen) [24] and AARAE (developed at the University of
responses) so that human decision making is removed. Sydney) [25]. The code for these was written independently.
Hence, some results may be inferior if there are cases for Both of these projects are freely available as open-source
which an expert might have optimised the analysis by chang- Matlab-hosted projects, aiming to support teaching and
ing the settings, by editing the impulse response, by choosing research in room acoustics and related areas. These projects
the most appropriate value to report or by not reporting a have a wide range of possible settings for reverberation time
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calculation, but the results reported in this paper are from responses with uniform octave band reverberation times in
the respective authors’ preferred settings. In both cases, this the 125–8000 Hz range.
included automatic end-truncation, compensating for decay
energy lost from truncation and background noise power sub- 3.2 Exponential Decay with Steady Noise Floor
traction. However, the implementations differ in details, such
as the filter-bank design, and the methods to find the trun- Apart from the relatively trivial task of calculating reverber-
cation point and to do the abovementioned adjustments. For ation time from a noise-free impulse response, probably the
the ITA Toolbox, the settings are exactly those described by most telling simple test is to introduce a steady-state noise
Guski and Vorländer [6] as ‘Method E’. The most distinc- floor. This means that automatic end-truncation must be done
tive feature of the chosen AARAE settings was that they by the analysis software, as well as associated adjustments.
applied the suggestion by Venturi et al. [21] for highly selec- An exponentially decaying envelope with a perfectly steady
tive octave band filtering (implemented as a reverse-time, or noise floor can be generated by adding a tonal noise term to
maximum phase, filter-bank in the frequency domain, with Eq. 1, as expressed by Eq. 2, where PNR is the peak-to-noise
144 dB/octave skirts). ratio expressed in decibels. Note that the tonal noise is in
quadrature with the tonal decay, so that they add similarly
to incoherent noises in terms of the resulting envelope, but
3 Tests and Results without random fluctuations. The noise term is multiplied by
the square root of 2 because PNR is a ratio of maximum to
3.1 Noise-Free Artificial Impulse Responses root-mean-square amplitude.
7
A simple and trivial test of reverberation time software y (t) = sin 2k π f 0 t e−3ln(10)t/Tk
k=1
performance is to calculate the reverberation time of an expo- √
nentially decaying waveform. To avoid random fluctuations + 2 × 10 P N R/20 cos 2k π f 0 t . (2)
in the envelope, we use exponentially decaying sinusoidal
waves. Equation 1 expresses the waveform, y, as a function Results from testing this are more interesting than without
of time, t (starting at 0 s and finishing at a duration suffi- noise, and they indicate that the tested software is generally
ciently long for reverberation time calculation). The lowest well behaved in dealing with steady noise. Figure 1 shows
tone frequency in the present study is f 0 = 125 Hz, and results for T30 with a PNR of 30, 35, 40 and 45 dB. In inter-
seven octaves spanning 125 Hz to 8 kHz are used (k = 1–7). preting this, it should be borne in mind that a signal-to-noise
In Eq. 1, Tk is the octave band reverberation time, and so the ratio of 45 dB or greater is required for standard calculation
error in the reverberation time calculation software can be of T30 , so the lower PNR values do not meet this require-
found by comparison to this. ment. Admittedly there is some subtlety in signal-to-noise
ratio metrics for impulse responses as explained by Hak et
7 al. [5], but in this paper we use PNR because of its simplic-
y (t) = sin 2k π f 0 t e−3ln(10)t/Tk . (1)
k=1
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will be less important in most real cases. However, the inter- because it results in a shift of energy backwards in time.
esting point here is that very high selectivity is achievable Hence zero-phase filtering is conceptually ideal for centre
without causing problems with the measurement of short time, but not ideal for reverberation time. The theoretical
reverberation times. To examine this further, a synthetic tonal centre time, Ts , of a simple exponential decay is proportional
impulse response with octave band reverberation times of to its reverberation time (Eq. 4).
0.01 s was tested with the AARAE 24th-order filter-bank
implementation. The measured result for T30 is 0.01 s in all T
Ts = . (4)
seven bands (using ‘double’ precision format for the input 6ln (10)
waveform, instead of the 16-bit signed integer wav file format
that was used for the other tests reported here). This confirms Tests using AARAE with ideal exponential decay of
the proposition by Venturi et al. [16] that a highly selective T = 1 s (for which Ts = 72.38 ms) using a time-reverse
filter-bank provides robust analysis, and it extends this by filter of order 6 to 24 yielded octave band centre time values
showing that the filter-bank does not introduce significant with errors no more than 0.1 ms, which can be neglected. A
time-related errors for the shortest conceivable reverberation shorter reverberation time is more vulnerable to filter effects,
time, at least for T30 calculation from an exponential decay. and a decay of T = 0.1 s (for which Ts = 7.238 ms) was
also tested with filter orders 6 to 24. In that case, the errors
3.3.1 Further Implications of Filter-Bank Design: Other were higher in absolute terms and much higher in relative
Parameters terms, with a maximum error of 0.27 ms. This error can still
probably be regarded as acceptable, especially since Ts is
In the present work we focus on reverberation time, but a unlikely to provide any insights to situations with very short
filter-bank ideal for that might not be ideal for another para- reverberation times. These maximum errors occurred in the
meter. For example, for speech transmission index (STI) 125 Hz band, and errors were an order of magnitude smaller
calculation, zero or linear phase filters are ideal because in high frequency bands. Errors did not increase as the fil-
phase distortion within each octave band must be min- ter order increased. Considering this, not using a zero-phase
imised to avoid changing the octave band envelopes [26]. filter-bank for octave band centre time appears to be unprob-
Furthermore, excessive filter order artificially reduces the lematic from a practical standpoint, at least for the bands
modulation transfer function [27]. Therefore, Cabrera et al. centred on 125 Hz and higher.
[26] selected a linear phase 12th-order octave band filter-bank Although the aim of this study is to examine T30 results of
for STI analysis. On the other hand, with regard to the clar- the surveyed instances of software, the considerations in this
ity and definition energy ratio parameters in ISO3382-1, the section of the paper naturally raise the question of software
filter phase response and order do not fundamentally affect performance for clarity index and centre time. Figure 3 shows
results (apart from controlling spectral leakage) because the the C50 and Ts errors from a noise-free synthetic impulse
relevant time periods are extracted prior to filtering, so a high- response with T = 1 s in all seven bands. For the most part,
order reverse-time filter can be used. However to avoid error, the errors in C50 are negligible, although one software imple-
the filters’ reverse-time decay should be captured, which may mentation (F) deviates by more than 1 dB in the 125 Hz band.
require zero-padding the relevant time periods extracted from On the other hand, Ts errors in excess of 5 % are common in
the waveform prior to filtering. The theoretical clarity index the lower two octave bands, which could be due to forward
of a simple exponential decay is known from Eq. 3, where temporal smearing by the octave band filters.
t E is the limiting time between early and late (0.05 s for C50 ,
0.08 s for C80 ), and T is reverberation time. 3.4 Realistic Measured Impulse Response
1 − e−t E 6ln10/T Real measured impulse responses are much more compli-
C1000t E = 10log10 . (3)
e−t E 6ln10/T cated than the synthetic waveforms tested here. While they
may raise multiple diverse challenges for analysis, they do
This was tested using AARAE, with a reverse-time not have known reverberation time values a priori, and so
octave band filter-bank spanning 125–8000 Hz. Exponen- there is not an exactly correct result for error analysis. We
tially decaying octave-spaced tones over a range of rever- use the same impulse response as Katz [10] in the present
beration times (0.1–2 s) yielded negligible errors (<0.2 dB) paper, more than 1 decade after Katz’s study. It should be
in the seven octave bands for C50 and C80 , and increasing noted that Katz allowed expert analysts to contribute to the
the filter order from 6 to 24 did not increase the error (in fact analysis—for example, they might manually choose the start
it slightly reduced it). and end times to prepare for the analysis. By contrast, the
Conceptually, centre time may be affected by reverse-time present study excluded pre-manipulation by a human expert,
filtering, especially at high orders in the low frequency bands, and relied entirely on the automatic processes in the software
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Apart from the 4 kHz band, the ITA Toolbox and AARAE
implementations return results that are essentially at the
median. Although AARAE returns an outlier at 4 kHz, its
deviation in that band returns only 0.01 s from the median.
Table 1 provides a comparison between the results of the
present study and those of Katz. The median values are sim-
ilar, but the standard deviations and interquartile ranges in
the present study are much smaller. The interquartile range
excludes outliers, such as implementations C and A in the
lower bands. This reduced spread of results in the present
study could be from a combination of factors, including
improvements in software implementations, the selection of
software, the human contribution to analysis in Katz’s study
(which was not in the present study) and statistical issues
relating to sample size. The end-truncation method in the
current standard (ISO3382-1-2009) provides less flexibility
than in the version at the time of Katz’s study (ISO3382-
1997) [28], which may have contributed to greater agreement
between implementations.
To put the statistical dispersion of results in context, the
last row of Table 1 shows the greatest absolute value devi-
Fig. 5 The upper chart shows octave band reverberation time (T30 ) ation from the median in the present study, expressed as a
calculated from the impulse response published by Katz [10]. The lower percentage. In other words, this represents the furthest out-
chart shows the deviation from median (i.e. the calculated value minus lier (either above or below the median), and provides a simple
the median value) in each octave band
way of comparing with the 5 % acceptable deviation rule of
thumb. This is not to say that the median is correct—it is
merely used as a reference for comparison. The PNR in the
125 Hz band is almost 45 dB, and is greater than this in the
ing to steady noise that was used for the synthetic test of PNR, other bands (see Fig. 4), and so the signal-to-noise ratio is
the real impulse response has fluctuations in both the decay essentially that required for T30 calculation. Of the nine soft-
and the noise floor. Hence the differences exhibited here by ware implementations, five had deviations from the median
B, C and F may be due to the sensitivities of their particular greater than 5 % in the 125 Hz band, three in the 250 Hz band,
algorithm used to identify the truncation point, and perhaps and one in the 500 Hz band.
the way in which the late decay rate or the noise floor values In comparing dispersion of T30 with the PNR in each
are calculated. The lower frequency selectivity of B’s filter- octave band, PNR appears to be a plausible predictor of vari-
bank might also contribute to its divergence from C and F, ation between the implementations’ results. Figure 6 shows
which is supported by the similarity of its results with those this comparison for the present study and for Katz’s study,
of G (which has similarly low selectivity) in the lower two with dispersion expressed in relative terms, as interquartile
bands. ratio (interquartile range divided by median). This concurs
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