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2
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3
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4
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- The desidered result is the linear impulse response of the acoustic
propagation h(t). It can be recovered by knowing the test signal x(t)
and the measured system output y(t).
- It is necessary to exclude the effect of the not-linear part K and of
the background noise n(t).
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- x(t) is a sine signal, which
frequency is varied exponentially with time, starting at f1
and ending at f2.
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- Not-linear behaviour of the loudspeaker causes many harmonics to appear
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- The deconvolution of the IR is obtained convolving the measured signal
y(t) with the inverse filter z(t) [equalized, time-reversed x(t)]
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- The “time reversal mirror” technique is employed: the system’s impulse
response is obtained by convolving the measured signal y(t) with the
time-reversal of the test signal x(-t). As the log sine sweep does not
have a “white” spectrum, proper equalization is required
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10
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- The last impulse response is the linear one, the preceding are the
harmonics distortion products of various orders
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11
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- After the sequence of impulse responses has been obtained, it is
possible to select and extract just one of them:
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12
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- A special plugin has been developed for the computation of STI according
to IEC-EN 60268-16:2003
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13
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- A special plugin has been developed for performing analysis of
acoustical parameters according to ISO-3382
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14
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- The new module is still under development and will allow for very fast
computation of the AQT (Dynamic Frequency Response) curve from within
Adobe Audition
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15
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16
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- Pre-ringing at high frequency due to improper fade-out
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- Perfect Dirac’s delta after removing the fade-out
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- Pre-ringing at low frequency due to a bad sound card featuring
frequency-dependent latency
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- The Kirkeby inverse filter is computed inverting the measured IR
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- Convolving the time-smeared IR with the Kirkeby compacting filter, a
very sharp IR is obtained
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- An anechoic measurement is first performed
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- A suitable inverse filter is generated with the Kirkeby method by
inverting the anechoic measurement
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- The inverse filter can be either pre-convolved with the test signal or
post-convolved with the result of the measurement
- Pre-convolution usually reduces the SPL being generated by the
loudspeaker, resulting in worst S/N ratio
- On the other hand, post-convolution can make the background noise to
become “coloured”, and hence more perciptible
- The resulting anechoic IR becomes almost perfectly a Dirac’s Delta
function:
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25
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26
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- Often a pulsive noise occurs during a sine sweep measurement
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- After deconvolution, the pulsive sound causes untolerable artifacts in
the impulse response
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- Several denoising techniques can be employed:
- Brutely silencing the transient noise
- Employing the specific “click-pop eliminator” plugin of Adobe Audition
- Applying a narrow-passband filter around the frequency which was being
generated in the moment in which the pulsive noise occurred
- The third approach provides the better results:
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- When the measurement is performed employing devices which exhibit
signifcant clock mismatch between playback and recording, the resulting
impulse response is “skewed” (stretched in time):
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- It is possible to re-pack the impulse response employing the
already-described approach based on the usage of a Kirkeby inverse
filter:
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- However, it is always possible to generate a pre-stretched inverse
filter, which is longer or shorter than the “theoretical” one - by
proper selection of the lenght of the inverse filter, it is possible to
deconvolve impulse responses which are almost perfectly “unskewed”:
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- When several impulse response measurements are synchronously-averaged
for improving the S/N ratio, the late part of the tail cancels out,
particularly at high frequency, due to slight time variance of the
system
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- However, if averagaing is performed properly in spectral domain, and a
single conversion to time domain is performed after averaging, this
artifact is significantly reduced
- The new “cross Functions” plugin can be used for computing H1:
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37
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- Analysis of performances of binaural dummy heads
- Analysis of performances of omni /
figure-of-8 microphone assemblies
- Polar patterns of dodechaedron loudspeakers
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38
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- The initial approach was to use directive microphones for gathering some
information about the spatial properties of the sound field “as
perceived by the listener”
- Two apparently different approaches emerged: binaural dummy heads and
pressure-velocity microphones:
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- It was attempted to “quantify” the “spatiality” of a room by means of
“objective” parameters, based on 2-channels impulse responses measured
with directive microphones
- The most famous “spatial” parameter is IACC (Inter Aural Cross
Correlation), based on binaural IR measurements
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40
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- Other “spatial” parameters are the Lateral Energy ratios: LE, LF, LFC
- These are defined from a 2-channels impulse response, the first channel
is a standard omni microphone, the second channel is a “figure-of-eight”
microphone:
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- Both IACC and LF depend strongly on the orientation of the microphones
- Binaural and pressure-velocity measurements were performed in 2 theatres
employing a rotating table for turning the microphones
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42
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- Experiment performed in anechoic room - same loudspeaker, same source
and receiver positions, 5 binaural dummy heads
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- 90° incidence - at low frequency IACC is almost 1, at high frequency the
difference between the heads becomes evident
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- Diffuse field - the difference between the heads is now dramatic
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- Experiment performed in the Auditorium of Parma - same loudspeaker, same
source and receiver positions, 5 pressure-velocity microphones
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- At 7.5 m distance, the results already exhibit significant scatter
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- At 25 m distance, the scatter is even larger....
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48
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49
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- LookLine D-300 dodechaedron
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50
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- LookLine D-200 dodechaedron
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51
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- Omnisonic 1000 dodechaedron
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- ESS is now employed in top-grade measurement systems, including Audio
Precision (TM), Rhode-Schwartz and Bruel & Kjaer’s DIRAC software
- However, these completely-packaged measurement systems often do not
allow to play “tricks” and to adjust the signals for solving problems,
which have been shown here
- Workarounds have been found for almost all the problems occurring when
performing ESS measurements
- These workarounds are easily applied by working with a general purpose
sound editor (Adobe Audition)
- A number of additional plugins have been developed, making easy
to generate the test signal, to deconvolve and process impulse
responses, to compute inverse filters and to perform advanced processing
(STI, AQT, etc.)
- These plugins are freely downlodable at the AURORA web site:
- www.aurora-plugins.com
- The only remaining problems are related to existing transducers
(microphones and loudspeakers), as their directivity is far from the
theoretical one
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