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Convention Paper
Presented at the 122nd Convention
2007 May 58 Vienna, Austria
The papers at this Convention have been selected on the basis of a submitted abstract and extended precis that have been peer
reviewed by at least two qualified anonymous reviewers. This convention paper has been reproduced from the author's advance
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ABSTRACT
Sine sweeps are employed since long time for audio and acoustics measurements, but in recent years (2000 and
later) their usage became much larger, thanks to the computational capabilities of modern computers. Recent
research results allow now for a further step in sine sweep measurements, particularly when dealing with the
problem of measuring impulse responses, distortion and when working with systems which are neither time
invariant, nor linear.
The paper presents some of these advancements, and provide experimental results aimed to quantify the
improvement in signal-to-noise ratio, the suppression of pre-ringing, and the techniques employable for performing
these measurements cheaply employing a standard PC and a good-quality sound interface, and currently available
loudspeakers and microphones.
lot of papers were published (particularly remarkable modified versions of the Aurora plugins [2]. Three
were the JAES papers of Muller/Massarani [5] and of rooms were chosen for the test: a small listening room
Embrechts et al. [6]). The tradeoffs of this technique equipped with a professional surround-sound
were understood much better, and it was recognized the monitoring system, a concert hall employing a wide-
need of further perfecting the measurement technique band, two-way dodechaedron loudspeaker, and the
for dealing with some problems. passenger's compartment of a car.
- pre-ringing at low frequency before the arrival of the
Various kinds of microphones were employed too, with
direct sound pulse
the goal of assessing if the measurement of certain
- sensitivity to abrupt pulsive noises during the acoustical quantities, such as the "spatial parameters"
measurement described in ISO 3382, and namely LF, LFC and IACC,
can be reliably measured with currently available top-
- skewing of the measured impulse response when the
brand microphones.
playback and recording digital clocks were mismatched
- cancellation of the high frequencies in the late part of The results show that, whilst some of the proposed
the tail when performing synchronous averaging methods really improve substantially the sine sweep
measurement method, solving the problems shown
- time-smearing of the impulse response when
above, on the other hand the weak part of the
amplitude-based pre-equalization of the test signal was
measurement chain is still about transducers, and
employed
namely loudspeakers and microphones, which do not act
always along our expectations, and which can cause
All of the problems pointed out here have been severe artifacts in the measured quantities.
investigated, and several solutions have been proposed.
It is therefore concluded that any impulse response
This paper presents these "refinements" to the original
measurement chain can be used with confidence only
exponential sine sweep technique, and divulgates the after a set of careful preliminary tests and alignments.
results of some experiments performed for assessing the Without this, the results are prone to be at least
effectiveness of these techniques.
suspicious, and significant errors have been found in the
experimental tests. Of consequence, it appears necessary
The methods analyzed include: to further improve the current measurements standards,
and mainly ISO 3382, for ensuring reliable and
- post-filtering of the time-reversal-mirror inverse filter reproducible measurements employing this (and other)
for avoiding pre-ringing methods of measuring impulse responses.
- "exact" deconvolution by division in frequency
domain with regularization
- counter-skewing of the measured impulse response This chapter is recalling the theory already presented in
when the playback and recording digital clocks are [1], so the reader has a consequential presentation of the
mismatched basic method, before discussing problems and
possible enhancements. The reader already knowing this
- employing running-time cross-correlation for method can skip directly to chapter 3.
performing proper synchronous averaging without
cancellation effects When spatial information is neglected (i.e., both source
and receivers are point and omnidirectional), the whole
The experiments for assessing the behavior of these information about the rooms transfer function is
"enhanced" measurement techniques were performed contained in its impulse response, under the common
employing a state-of-the-art hardware system, including hypothesis that the acoustics of a room is a linear, time-
a multichannel sound interface, a powerful PC, and invariant system.
This includes both time-domain effects (echoes, discrete The quantity which we are initially interested to
reflections, statistical reverberant tail) and frequency- measure is the impulse response of the linear system
domain effects (frequency response, frequency- h(t), removing the artifacts caused by noise, not-linear
dependent reverberation). behavior of the loudspeaker and time-variance.
The following figure shows how a room can be seen, The method chosen, based on an exponential sweep test
under these hypotheses, as a single-input, single-output signal with aperiodic deconvolution, provides a good
black box. answer to three above problems: the noise rejection is
better than with an MLS signal of the same length, not-
linear effects are perfectly separated from the linear
Noise n(t) response, and the usage of a single, long sweep (with no
synchronous averaging) avoids any trouble in case the
system has some time variance.
It must also be taken into account the fact that the test
signal has not a white (flat) spectrum: due to the fact
that the instantaneous frequency sweeps slowly at low
frequencies, and much faster at high frequencies, the
resulting spectrum is pink (falling down by -3 dB/octave
in a Fourier spectrum). Of course, the inverse filter must
compensate for this: a proper amplitude modulation is
consequently applied to the reversed sweep signal, so
that its amplitude is now increasing by +3 dB/octave, as
shown in fig. 5.
h ( t ) = y( t ) f ( t ) (2)
The advantage of the new technique above the Despite the significant advantages shown by the ESS
traditional MLS method can be shown easily, repeating method in comparison with all the other previously-
the measurement in the same conditions and with the employed methods, some problems can still be found, as
very same equipment. Fig. 7 shows this comparison in already pointed out in chapter 1.
the case of a measurement made in a highly reverberant
space (a church). In the following subchapters, each of these problems is
analyzed, and proper workarounds are presented.
It is easy to see how the exponential sine sweep method
produces better S/N ratio, and the disappearance of 3.1. Pre-ringing
those nasty peaks which contaminate the late part of the
MLS responses, actually caused by the slew rate The measured impulse response often shows some
limitation of the power amplifier and loudspeaker significant pre-ringing before the arrival of the direct
employed for the measurements, which produce severe sound.
harmonic distortion.
However, the situation ameliorates significantly if we The following figure shows the result of a loopback
remove the fade-out. The following figure show the measurement, employing the same parameters as for the
results obtained with exactly the same settings as in the previous example (fs=44100 Hz, sweep from 22 Hz to
previous case, but with a length of the fade-in set to 0.0s 22050 Hz, 15s long, 0.1s fade-in, no fade-out).
(fade-in is still 0.1s).
f f
est
A packing filter is a filter capable of compacting the Fig. 11 frequency-dependent regularization parameter
time-signature of the impulse response. Various
methods for creating a numerical approximation to an The following figure shows the inverse filter computed
ideal packing filter have been proposed in the past. The for compacting the loopback IR shown in fig. 10:
method employed here is the one developed by Ole
Kirkeby, when working at the ISVR with prof. Nelson
[9]. Although Kirkeby did propose this method for
multichannel inversion (cross-talk cancellation), it can
be successfully employed also just for the purpose of
packing in time the transfer function of a single-input,
single-output system.
Then, as usual, a reference anechoic measurement is After convolution with the inverse filter, this pulsive
performed (employing the pre-equalized test signal); a event causes a quite evident artifact on the deconvolved
Kirkeby inverse filter is thereafter computed, with the IR, as shown here:
goal of removing the residual colouring of the
measurement system. This inverse filter is applied as a
post-filter, to the measured data, ensuring that the total
transfer function of the measurement system is made
perfectly flat. This is the approach successfully
employed in the Waves project, as described in more
detail in [8].
3.3. Pulsive noises during the measurement Fig. 21 Artifact caused by a pulsive event
When long sweeps are employed for improving the In practice, the artifact is a sort of frequency-decreasing
signal-to-noise ratio, the risk that some pulsive noise sweep, starting well before the beginning of the linear
occurs during the measurement increases, as it is impulse response, and continuing after it. The first part
difficult to keep people perfectly still for more than a is practically irrelevant on the linear IR, as it will be cut
few seconds. Typical sources of pulsive noise are away together with the harmonic distortion responses.
objects falling on the floor, seats being moved, or
cracks caused by steps over wooden floors. However, the part of this spurious sweep occurring in
the late part of the measurement can cause severe
The following sonogram shows a recorded sweep problems. In particular, when analyzing the reverberant
contaminated by an evident spurious pulsive event (the tail, this artifact is causing large errors on the estimate
vertical line), caused by an object falling on the floor. of the reverberation time and of the other acoustical
parameters computed according to ISO 3382. The
following figure shows a comparison between the
octave-band-filtered IR with and without contamination
by the spurious pulsive noise.
A much better removal of the pulsive event is obtained Fig. 27 effect of the pulsive event
by employing the Click/Pop Eliminator provided by on the deconvolved IR after click/pop Eliminator
Adobe Audition. The following picture shows how it
works: The artifact has been further reduced, but it is still there.
In this case, the result of the deconvolution is the Fig. 28 usage of FFT Filter for removing the pulsive
following: artifact
Various methods can be applied for re-aligning the Nevertheless, this approach requires the availability of a
clocks. For example, if a reference measurement can clean reference measurement, performed either
be performed, we could try to use a Kirkeby inverse electrically (as in this example) or under anechoic
filter for fixing the mismatch, as already shown in conditions.
chapters 3.1 and 3.2.
Whenever a reference measurement is not available, the
The following figure show the result of such an inverse inverse filter approach cannot be employed. Another
filter applied to the electrical measurement performed. possible solution is the usage of a pre-strecthed inverse
filter for performing the IR deconvolution.
sweep, and its inverse sweep, 8.5 ms longer than the 3.5. Time averaging
original one.
The usage of averaging several impulse responses for
When we convolve this longer inverse sweep with the improving the signal-to-noise ratio is a deprecated
recorded signal, the deconvolution produces the technology when working with the ESS method.
following result:
Synchronous time averaging works only if the whole
system is perfectly time-invariant. This is never the case
when the system involves propagation of the sound in
air, due to air movement and change of the air
temperature. So, the preferred way for improving the
signal to noise ratio is not to average a number of
distinct measurements, but instead to perform a single,
very long sweep measurement, as clearly recommended
in the ISO 18233/2006 standard.
Fig. 34 correction of a skewed measurement The following picture compares the sonographs of two
employing deconvolution with a longer inverse sweep IRS, the first comes from a single, long sweep of 50s,
the second from the average of a series of 50 short
This result is not so clean as the one obtained with the sweeps of 1s each.
Kirkeby inversion, but now we have got a quite good
clock realignment without the need of a reference
measurement.
Fig. 36 spectrum of single sweep of 50s (above) It can be seen how the single-sweep measurement is
versus 50 sweeps of 1s (below) providing a perfectly linear decay with quite good
dynamic range (63 dB), whilst the synchronously-
It can be seen how, above 350 Hz, the synchronously- averaged IR exhibit strong underestimate of the energy
averaged IR is systematically underestimated. Around of the reverberant tail, and simultaneously a much worst
5-6 kHz the underestimation is more than 10 dB. signal-to-noise ratio (43 dB).
This of course affects also the slope of the decay curve, It can be concluded that synchronously-averaging a
and the estimate of reverberation times. The following number of subsequent IRs obtained with the ESS
figure shows the comparison between the octave-band method is causing unacceptable artifacts.
filtered impulse response and decay curves at 4 kHz:
However, an alternative technique can be used, in these
cases, for processing the data.
H1 (f ) =
G LR (5) Analyzing the octave-band-filtered impulse response (at
G LL 4 kHz), the following is obtained:
4. PERFORMANCE OF ELECTROACOUSTIC
TRANSDUCERS
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in anechoic conditions for three dodechaedrons. The Fig. 44 directivity patterns at 2 kHz
first one is a standard-size (40cm diameter) employing
for building acoustics measurements (LookLine D-300); Horizontal Polar Plot - LookLine D300 - 4000 Hz
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Horizontal Polar Plot - LookLine D200 - 4000 Hz
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Horizontal Polar Plot - Omnisonic - 4000 Hz
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The following figure shows the three dodechaedrons Fig. 45 directivity patterns at 4 kHz
analyzed:
It can be seen how all three these dodecaedrons exhibit
quite irregular polar patterns at medium-high frequency.
1
Comparison LF - measure 2 - 25m distance It can be seen that, even at medium frequencies, the
0.9
Schoeps figure-of-8 pattern is distorted, and is not properly gain-
Neumann
0.8 Soundfield
matched with the omnidirectional one. These deviations
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cross correlation), also defined in ISO-3382, and
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However, various makers of dummy heads produce
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quite different microphone assemblies. For checking
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comparatively their performances, a set of impulse
response measurements have been performed in a large
Fig. 49 ST-250 polar patterns at 500 Hz and 2 kHz
anechoic chamber, employing a turntable controlled by
the sound card, as shown in the following figure:
5. ACKNOWLEDGEMENTS
6. REFERENCES
[1] A.Farina Simultaneous measurement of impulse
response and distortion with a swept-sine
technique, 110th AES Convention, February 2000.
Fig. 51 anechoic measurements on dummy heads
[2] www.aurora-plugins.com
Also in this case 4 different binaural microphones have [3] P.Craven, M.Gerzon - "Practical Adaptive Room
been tested: And Loudspeaker Equaliser for Hi-Fi Use" - 92nd
Bruel & Kjaer type 4100 AES Convention, March 1992
Cortex [4] D.Griesinger - "Beyond MLS - Occupied Hall
Measurement With FFT Techniques" - 101st AES
Head Acoustics HMS-III Convention, Nov 1996
Neumann KU-100 [5] S. Mller, P. Massarani Transfer-Function
Measurement with Sweeps, JAES Vol. 49,
A synthetic diffuse sound field has been generated, Number 6 pp. 443 (2001).
employing a number of loudspeakers surrounding the
dummy head and feeding them with uncorrelated pink [6] G. Stan, J.J. Embrechts, D. Archambeau
noise. Comparison of Different Impulse Response
Measurement Techniques, JAES Vol. 50, No. 4, p.
In principle, given the fact that the sound field was 249, 2002 April.
exactly the same, all the dummy heads should have [7] A. Torger, A. Farina Real-time partitioned
given the same value of IACC. Instead, as shown in the convolution for Ambiophonics surround sound,
following figure, the results have been quite diverging: 2001 IEEE Workshop on Applications of Signal
Processing to Audio and Acoustics - Mohonk
IACCe - random incidence
Mountain House New Paltz, New York October 21-
1
24, 2001.
0.9
Cortex
0.5
Head
Neumann
2003
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