Notes
Slide Show
Outline
1
Advancements in impulse response measurements by sine sweeps
2
Topics
3
Basic sound propagation scheme
4
Measurement process
  • 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).


5
Exponential Sine Sweep method
  •  x(t) is a sine signal, which frequency is varied exponentially with time, starting at f1 and ending at f2.
6
Test Signal – x(t)
7
Measured signal - y(t)
  • Not-linear behaviour of the loudspeaker causes many harmonics to appear
8
Inverse Filter – z(t)
  • The deconvolution of the IR is obtained convolving the measured signal y(t) with the inverse filter z(t) [equalized, time-reversed x(t)]
9
Deconvolution of Log Sine Sweep
  • 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


10
Result of the deconvolution
  • The last impulse response is the linear one, the preceding are the harmonics distortion products of various orders
11
IR Selection
  • After the sequence of impulse responses has been obtained, it is possible to select and extract just one of them:
12
Post processing of impulse responses
  • A special plugin has been developed for the computation of STI according to IEC-EN 60268-16:2003
13
Post processing of impulse responses
  • A special plugin has been developed for performing analysis of acoustical parameters according to ISO-3382
14
The new AQT plugin for Audition
  • 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
15
Topics
16
Pre-ringing at high and low frequency
  • Pre-ringing at high frequency due to improper fade-out
17
Pre-ringing at high and low frequency
  • Perfect Dirac’s delta after removing the fade-out
18
Pre-ringing at high and low frequency
  • Pre-ringing at low frequency due to a bad sound card featuring frequency-dependent latency
19
Kirkeby inverse filter
  • The Kirkeby inverse filter is computed inverting the measured IR
20
Pre-ringing at high and low frequency
  • Convolving the time-smeared IR with the Kirkeby compacting filter, a very sharp IR is obtained
21
Topics
22
Equalization of the whole system
  • An anechoic measurement is first performed
23
Equalization of the whole system
  • A suitable inverse filter is generated with the Kirkeby method by inverting the anechoic measurement
24
Equalization of the whole system
  • 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:
25
Topics
26
Sensitivity to abrupt pulsive noises
  • Often a pulsive noise occurs during a sine sweep measurement
27
Sensitivity to abrupt pulsive noises
  • After deconvolution, the pulsive sound causes untolerable artifacts in the impulse response
28
Sensitivity to abrupt pulsive noises
  • 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:
29
Topics
30
Clock mismatch
  • 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):
31
Clock mismatch
  • It is possible to re-pack the impulse response employing the already-described approach based on the usage of a Kirkeby inverse filter:
32
Clock mismatch
  • 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”:
33
Topics
34
High-frequency cancellation due to averaging
  • 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
35
High-frequency cancellation due to averaging
  • 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:
36
Topics
37
Directivity of transducers
  • Analysis of performances of binaural dummy heads


  • Analysis of performances of omni /
    figure-of-8 microphone assemblies


  • Polar patterns of dodechaedron loudspeakers


38
Spatial analysis by directive microphones
  • 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:


39
“objective” spatial parameters
  • 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


40
“objective” spatial parameters
  • 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:



41
Robustness of spatial parameters
  • 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
42
Are binaural measurents reproducible?
  • Experiment performed in anechoic room - same loudspeaker, same source and receiver positions, 5 binaural dummy heads
43
Are binaural measurents reproducible?
  • 90° incidence - at low frequency IACC is almost 1, at high frequency the difference between the heads becomes evident
44
Are binaural measurents reproducible?
  • Diffuse field - the difference between the heads is now dramatic
45
Are LF measurents reproducible?
  • Experiment performed in the Auditorium of Parma - same loudspeaker, same source and receiver positions, 5 pressure-velocity microphones
46
Are LF measurents reproducible?
  • At 7.5 m distance, the results already exhibit significant scatter
47
Are LF measurents reproducible?
  • At 25 m distance, the scatter is even larger....
48
Directivity of transducers
49
Directivity of transducers
  • LookLine D-300 dodechaedron
50
Directivity of transducers
  • LookLine D-200 dodechaedron
51
Directivity of transducers
  • Omnisonic 1000 dodechaedron
52
Conclusions
  • 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