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Applied Acoustic - test of 14/12/2021
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Matricula
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1)     Check the sentences you think are always TRUE
(multiple answers allowed)
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¨  IIR filters are always preferable to FIR filters, as IIR is more efficient computationally
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¨  The design of IIR filters is tricky and difficult, as they can be unstable
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¨  The design of FIR filters is simple, and they are always stable
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¨  A modern processor can use FIR filters up to a few thousands taps
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¨  A modern processor can use FIR filters of any length (even millions of taps)
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¨  Aliasing occurs when an analog-to-digital converter is operated with a limited resolution (less than 24 bits)
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2)     An Ambisonics multichannel audio track can be:
(multiple answers allowed)
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¨  recorded using a compact microphone array
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¨  synthesized from a number of mono tracks, spatially panned in different locations
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¨  reproduced over headphones by convolving with a proper binaural filter matrix
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¨  reproduced over a loudspeaker rig, employing a proper decoder
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¨  converted to any other "coincident" stereo recording, such as XY or MS
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¨  converted to any closely-spaced stereo recording, such as ORTF or INA
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¨  employed for feeding a Dolby Digital Surround system (5.1)
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3)     What is "spatial aliasing" ?
(a single answer)
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¡  It is the appearance of spurious peaks when the signal contains frequencies above the Nyquist frequency (half of the sampling frequency)
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¡  It is the appearance of the last part of an impulse response before the arrival of the direct sound, being folded back at the beginning when the IR was cut too short
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¡  It is the distortion of the spatial information occurring when the signal contains frequencies too large in comparison of the spacing between the microphones constituting a microphone array
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¡  It is the distortion of the spatial information occurring when the signal contains frequencies too large in comparison of the spacing between the loudspeakers in a WFS array
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¡  It is the disruption of the signal integrity caused by jitter on the sampling clock
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¡  It is the distortion of the waveform caused by using a converter with a limited resolution (number of bits)
Sx
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4)     On ORTF stereo microphone pair is recording a sound source located at an azimuth of 30+2*F degrees from front. Compute the inter-channel time delay.Alpha =42°
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(write number and measurement unit)
Az =42°
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b = d*sin(Alpha) =
0.1137522031mab
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DeltaTau = b/c =
0.3345653032ms
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17 cm
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5)     An Ambisonics microphone is recording a sound source located at an azimuth of 30+2*F degrees from front. Compute the level difference between channels X and Y.
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(write number and measurement unit)
Az =42°
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X = cos(alpha) =
0.7431448255
Delta(dB) = 20*log(X/Y) =
0.9112512594dB
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Y = sin(alpha) =
0.6691306064
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6)     A microphone is placed 20+E cm in front of a reflecting wall. The sound is impinging on the microphone perpendicularly to the wall surface, and bounces back to the microphone, causing a nasty comb filtering effect. Compute the frequency of the first notch.
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(write number and measurement unit)
d =0.25m2d =0.5
m ; Datau = 2d/c =
0.001470588235s
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d = Lambda/4
70.58823529samples
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Lambda = 4*d =
1m=c/f
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f = c/lambda =
340Hz
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7)     The sound system of a car has two speakers, and must be optimized for the driver. The distances of the two loudspeakers from the center of the driver's head are: rleft = 0.4+F/20 m, rright = 0.8 +E/25 m. Compute the delay to be applied to the left channel for aligning temporally the two signals.
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(write number and measurement unit)
rleft =0.7mrright =1m
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Delta-dist = rright-rleft =
0.3m
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Delta Tau = Delta-dist/c =
0.0008823529412s
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 0.8823529412ms
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8)     In the same case as the previous exercise, compute the gain reduction to be applied to the left channel for aligning the amplitude of the two direct-sound signals.
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(write number and measurement unit)
Lw =100dB
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Ll = Lw -11 -20*log10(rleft) =
92.0980392dB
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Lr = Lw -11 -20*log10(rright) =
89dB
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Gain Reduction Left =
3.0980392dB
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