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1
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- Angelo Farina and Lamberto Tronchin
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2
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3
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4
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5
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6
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9
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11
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12
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13
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21
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23
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24
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- We are interested in the linear impulse response h(t). This can be
estimated by the knowledge of the input signal x(t) and of the output
signal y(t).
- The influence of the not-linear part K and of the noise n(t) has to be
minimized.
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25
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- X(t) is a periodic binary signal obtained with a suitable
shift-register, configured for maximum lenght of the period.
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26
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- The re-recorded signal y(i) is cross-correlated with the excitation
signal thanks to a fast Hadamard transform. The result is the required
impulse response h(i), if the system was linear and time-invariant
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27
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28
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29
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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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30
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31
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- The not-linear behaviour of the loudspeaker causes many harmonics to
appear
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32
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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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33
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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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34
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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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35
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- It is possible to measure impulse responses in various formats:
- Mono (Omnidirectional)
- Stereo (ORTF)
- Binaural (Dummy Head)
- B-format (1st order Ambisonics, Soundfield microphone)
- WFS (Wave Field Synthesis, circular array)
- M. Poletti high-order virtual microphones
- Employing a multichannel sound card, all of these measurements can be
performed simultaneously
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36
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- Test Signal: pre-equalized sweep
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37
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- Equalized, omnidirectional sound source:
- Dodechaedron for mid-high frequencies
- One-way Subwoofer (<120 Hz)
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38
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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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39
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- 3 types of microphones:
- 2 Cardioids in ORTF placement (Neumann K-140)
- Binaural dummy head (Neumann KU-100)
- B-Format 4 channels (Soundfield ST-250)
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40
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41
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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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42
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43
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44
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45
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46
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47
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48
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49
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50
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51
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52
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53
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54
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55
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56
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- Stereo (ORTF on 2 standard loudspeakers at +/- 30°)
- Binaural on headphones
- Binaural on loudspeakers (Stereo Dipole)
- 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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57
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58
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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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59
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60
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61
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62
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- Step 1 – pass to frequency domain through FFT
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63
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64
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- System’s impulse response
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65
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- Convolution of inverse filter with the system’s impulse response
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66
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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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67
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68
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69
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70
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71
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- Reproduction occurs over an array of 8-24 loudspeakers, through an
Ambisonics decoder
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72
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73
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74
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75
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76
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77
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78
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79
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- One of the two ORTF cardioid is employed, which samples 36 positions
along a 110 mm-radius circumference
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80
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- Flow diagram of the process
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81
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- Stereo Dipole + Virtual Ambisonics
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82
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83
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84
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85
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- The experimental setup allows for measurements of high-quality mono,
binaural and B-format IRs
- A proper listening room is required in order to reproduce sound field
with Stereo Dipole and/or Ambisonic methodology
- The sound quality of different theatres can be assessed in real-time in
the listening room
- Questionnaires can be collected through an interactive-driven software
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86
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- Thanks to novel zero-latency convolution software, musicians can play a
keyboard and listen to a virtual spatial sound environment in real time.
- Spatial auralisation can be imediately switched or morphed while the
musician plays the keyboard.
- This technology has been made avaliable also for processing music in
recording studios, thanks to plugins developed by Sony, Waves, Voxengo,
Tascam, Altiverb
- The next step will be to port this “sampled reverberation” method also
for live applications
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Notes
