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- Angelo Farina (1) – Regev Ayalon (2)
- (1) Dipartimento di Ingegneria Industriale, Università di
Parma, Via delle Scienze 181/A
- Parma, 43100 ITALIA
- HTTP://pcfarina.eng.unipr.it -
mail: [email protected]
- (2) K.S. Waves Inc., Azrieli Center, Tel Aviv, ISRAEL
- HTTP://www.waves.com - mail:[email protected]
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- The title of this paper is exactly the same employed by Michael Gerzon
in its JAES paper (Vol. 23, Number 7, 1975)
- He first proposed to collect impulse responses measured in famous
theatres, with a microphone capable of capturing the complete spatial
information
- This paper is consequently basically a tribute to M.Gerzon, who had
foreseen most of the modern multichannel audio applications, including
impulse response measurements and auralization obtained by convolution.
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- The main goal is to measure an huge collection of impulse response in
famous theatres, concert halls, cathedrals, etc.
- These impulse responses have two main uses:
- In case something happens to the original space (remember the case of La
Fenice theater) they contain a detailed “acoustical photography” which
is preserved for the posterity
- They can be used for studio sound processing, as artificial reverb and
surround filters for today’s and tomorrow’s musical productions
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- Description of the measurement technique
- Analysis of some acoustical parameters of the first theaters already
measured
- Description of the processing methods to be employed for transforming
the measured data in audible reconstructions of the original spaces
- Description of the usage of the measured data for studio processing and
production
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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 variable exponentially with
time, starting at f1 and ending at f2.
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- The “time reversal mirror” technique is emplyed: 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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- The not-linear behaviour of the loudspeaker causes many harmonics to
appear
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- The deconvolution of the system’s impulse response is obtained
convolving the measured signal y(t) with the inverse filter z(t)
[equalized, time-reversed x(t)]
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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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- The measurement method incorporates all the known techniques:
- Binaural
- B-format (1st order Ambisonics)
- WFS (Wave Field Synthesis, circular array)
- ITU 5.1 surround (Williams MMA, OCT, INA, etc.)
- Binaural Room Scanning
- M. Poletti high-order virtual microphones
- This measurement setup has been named “Waves2003”, as it is being
employed for the collection of impulse response to be employed together
with the new convolution software being developed by KS Waves ltd.
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- Test Signal: pre-equalized sweep
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- Equalized, omnidirectional sound source:
- Dodechaedron for mid-high frequencies
- Subwoofer
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- Genelec S30D reference studio monitor:
- Three-ways, active multi-amped, AES/EBU
- Frequency range 37 Hz – 44 kHz (+/- 3 dB)
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- 3 types of microphones:
- Binaural dummy head (Neumann KU-100)
- 2 Cardioids in ORTF placement (Neumann K-140)
- B-Format 4 channels (Soundfield ST-250)
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- A single measurement session play backs 36 times the test signal, and
simultaneusly record the 8 microphonic channels
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- The basic method consists in convolution of a dry signal with a set of
impulse responses corresponding to the required output format for
surround (2 to 24 channels).
- The convolution operation can nowadays be implemented very efficiently
on a modern PC through an ancient algorithm (equally-partitioned FFT
processing, Stockam 1966).
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- Stereo (ORTF on 2 standard loudspeakers at +/- 30°)
- Rotation-tracking reproduction on headphones (Binaural Room Scanning)
- Full 3D Ambisonics 1st order (decoding the B-format signal)
- ITU 5.1 (from different 5-mikes layouts)
- 2D Ambisonics 3rd order (from Mark Poletti’s circular array
microphone)
- Wave Field Synthesis (from the circular array of Soundfield microphones)
- Hybrid methods (Ambiophonics)
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- Playback occurs over a pair of loudspeakers, in the standard
configuration at angles of +/- 30°, each being fed by the signal of the
corresponding microphone
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- Reproduction occurs over 2 loudspeakers angled at +/- 10°, being fed
through a “cross-talk cancellation” digital filtering system
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- Reproduction occurs over an array of 8-24 loudspeakers, through an
Ambisonics decoder
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- One of the two ORTF cardioid is employed, which samples 36 positions
along a 100 mm-radius circumference
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- Flow diagram of the process
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- Ambiophonics 3D (10 loudspeakers):
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- Main advantages of the new measurement method “Waves 2003”:
- Almost all previously known measurement techniques are incorporated in
a single, coherent approach
- The spatial informations are accurately sampled, making it possible to
store, analyze and preserve these “3D acoustical photographies” of
existing musical spaces for the posterity
- The impulse response are stored in many different formats, allowing for
their usage for surround productions with today technlogies (ITU 5.1,
1st order Ambisonics) and future, more advanced methods (high order
Ambisonics, WFS, Ambiophonics)
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