|The Science of Domestic Concert Hall Design|
by Ralph Glasgal
AES 24th International Conference on Multichannel Audio 1
Concert Hall Acoustics For Posterity
1 MEASUREMENT METHOD
This chapter describes the details of the measurement method, the equipment (hardware and software), and the procedure.
Although most of these items are not inherently new, the combination of them in a coherent approach provides a general method from which all known multichannel formats can be derived.
1.1 Test signal and deconvolution
The excitation-deconvolution technique employed for the measurement of the impulse response is the log sine sweep method, as initially suggested by one of the authors . Independent evaluations have shown that this method is superior to the previously employed ones [10, 11].
A good compromise between measured frequency range, length of the sweep and signal-to-noise ratio has been reached, by choosing the following parameters:
The ýunusualţ length of the silence between sweeps is due to the traveling time of the rotating table. The rotation is triggered by a proper pulsive signal, automatically generated in the middle of the silence gap on the second channel of the sound card.
The choice of the above parameters allows for measurement of impulse responses which have wide frequency span, good dynamic range (approximately 90 dB) and are substantially immune from background noise eventually present during the measurements. The deconvolution is obtained by linear (not circular) convolution with a proper inverse filter, which is automatically generated together with the test signal. As explained in , this inverse filter is simply the time reversal of the test signal, properly amplitude-equalized for compensating the 6 dB/oct falloff caused by the log sweep.
The linear deconvolution is effective in avoiding that not-linear behavior of the transducers can cause harmonic distortion artifacts affecting the measured impulse response.
As the playback-recording is performed at 96 kHz-24 bits, there is enough distance between the maximum generated frequency and the Nyquist frequency, that the ringing of the anti-aliasing filters is not excited, and the measured impulse response does not suffer from highfrequency phase distortion.
Also the amplitude of the emitted test signal has been properly amplitude-equalized, for compensating the uneven frequency response of the loudspeaker: this way, the emitted sound power has a reasonably flat spectrum over the whole frequency range.
Figs. 2 and 3 show respectively the equalized test signal (CoolEditPro was employed for playback & recording) and the userÝs interface of the software employed for the deconvolution. Thanks to the usage of the new, highly optimized Intel Integrated Performance Primitives v. 3.0 FFT routines, the deconvolution is now incredibly fast (approximately 20% of the duration of the recorded signal).
1.2 The sound source
An omnidirectional sound source is usually preferred for measurements of room impulse responses. Albeit this does not correspond to the effective directivity pattern of real-world sound sources (such as musical instruments or human talkers and singers), the usage of an omnidirectional sound source is predicated by current standards (ISO3382, for example), and avoids XX exploiting strange room effects, as can happen employing highly directive loudspeakers (abnormal energization of echoes and focalizations for selected orientations of the source).
A special, ultra-compact dodechaedron loudspeaker was built specifically for the purpose of this research, employing 12 full-range drivers installed on a small size enclosure (approx. diameter is 200 mm). This unit, of course, is not capable of producing significant acoustical power under 120 Hz; for extending the low frequency range a subwoofer was added, incorporating it inside the cylindrical transportation case, which also contains the power amplifier (300 W RMS) and serves as supporting base for the dodechaedron.
Fig. 4 shows a photograph of this special omnidirectional sound source.
The acoustical performance of the loudspeaker was measured inside an anechoic room, averaging the radiated sound over a complete circumference. As the 1/3 octave spectrum measured when feeding the loudspeaker with perfectly flat pink noise was significantly uneven, a proper equalization of the test signal was necessary. Fig. 5 shows the comparison between the radiated sound power of the loudspeaker prior and after the equalization, which was performed applying directly to the test signal the graphical 1/3 octave filtering required for flattening the response.
From the graph, it can be seen how the digital equalization was capable of flattening perfectly the loudspeakerÝs response between 80 and 16000 Hz, with a gentle roll-off outside this interval. After the equalization, the total radiated sound power level (with pink noise) was approximately 97 dB.