Authors:Thuillier, Etienne; Lähdeoja, Otso; Välimäki, Vesa
Affiliation:Aalto University, Acoustics Lab, Department of Signal Processing and Acoustics, Espoo, Finland; University of the Arts Helsinki, Center for Music & Technology, Helsinki, Finland; Aalto University, Acoustics Lab, Department of Signal Processing and Acoustics, Espoo, Finland
Recent research demonstrated that the classical guitar can be advantageously augmented using a pickup to drive an actuator mounted on the guitar’s back plate. This allows enrichment of the instrument’s timbral palette with audio effect processors in the loop. The feedback problem that results from such a setup is similar to what occurs in live music performance setups where the sound of a guitar is amplified using a loudspeaker. In the present case, measurements of the augmented guitar’s open-loop response demonstrate that instabilities are susceptible to occurring from the string’s modes and not from the guitar’s sound-box. In particular, the shape of the magnitude response suggests frequency shifting as a viable solution to string instability. Introduction of an upward frequency shift in the forward path is proposed as a means for stabilizing the closed-loop system. Experimental results demonstrate that the proposed solution leads to improved stability even for a modest frequency shift of 3 Hz. The achieved gain margin improvement, which is shown to be of at least 3 dB, then comes at the cost of a clearly perceptible amplitude modulation, which may be acceptable in conjunction with other audio effects chosen by the performer.
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Authors:Xie, Bosun; Mai, Haiming; Rao, Dan; Zhong, Xiaoli
Affiliation:State Key Laboratory of Subtropical Building Science, South China University of Technology, Guangzhou, China; Acoustic Lab., School of Physics and Optoelectronics, South China University of Technology, Guangzhou, China; Acoustic Lab., School of Physics and Optoelectronics, South China University of Technology, Guangzhou, China; Acoustic Lab., School of Physics and Optoelectronics, South China University of Technology, Guangzhou, China
In multichannel sound reproduction, recreating virtual sources or sound images in different directions can be realized with the principle of summing localization of multiple loudspeakers or real sound sources. On the basis of Wallach’s hypothesis that the variations in the interaural time difference caused by head turning provide dynamic cues for front-back and vertical localization, the present study develops a framework for analyzing the vertical summing localization of multichannel sound reproduction with amplitude panning. The previously derived localization equations, which were based on the simplified shadow-less head model, are reviewed and psychoacoustics explanations are provided. A HRTF-based method for analyzing the vertical summing localization more strictly is described. Based on the proposed framework and method, vertical summing localizations for pair-wise amplitude panning in the median plane and Ambisonics are analyzed. The results confirm previous observations that for some appropriate loudspeaker configurations, pair-wise amplitude panning is able to recreate a virtual source between two loudspeakers in the median plane.
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Authors:Taylor, Richard; Manke, Kurtis; Keele, D.B. (Don)
Affiliation:Thompson Rivers University, Kamloops, Canada; University of Victoria, Victoria, Canada; DBK Associates and Labs, Bloomington, IN, USA
This report develops the theory for a constant-beamwidth transducer formed by an unbaffled, continuous circular-arc isophase line source. Appropriate amplitude shading of the source distribution leads to a far-field radiation pattern that is constant above a cutoff frequency. If the active part of the array is limited to an arc of 180° or less, the radiation pattern is asymptotically frequency-independent above a cutoff frequency determined by the arc radius and the highest-order nonnegligible shading mode. The cutoff frequency is inversely proportional to the arc radius and prescribed beam width. Two shading functions are derived with cosine and Chebyshev polynomial forms, optimized to minimize this cutoff frequency and thereby extend constant-beamwidth behavior over the widest possible band. The authors illustrate the theory with simulations of magnitude responses, full-sphere radiation patterns, and directivity index for example arrays with both wide- and narrow-beam radiation patterns. The theory is extended to describe the behavior of circular-arc arrays of discrete point sources. The conclusions that follow are remarkably parallel to those for an amplitude-shaded spherical cap as developed in previous research.
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Authors:Geronazzo, Michele; Peruch, Enrico; Prandoni, Fabio; Avanzini, Federico
Affiliation:Dept. of Architecture, Design, and Media Technology, Aalborg University, Copenhagen, Denmark; Dept. of Information Engineering, University of Padova, Padova, Italy; Dept. of Information Engineering, University of Padova, Padova, Italy; Dept. of Computer Science, University of Milano, Milano, Italy
This paper describes an image-guided HRTF selection procedure that exploits the relation between features of the pinna shape and HRTF notches. Using a 2D image of a user’s pinna, the procedure selects from a database the HRTF set that best fits the anthropometry of that user. The proposed procedure is designed to be quickly applied and easy to use for a user without previous knowledge of binaural audio technologies. The entire process is evaluated by means of an auditory model for sound localization in the mid-sagittal plane available from previous literature and a short localization test in virtual reality. Using both virtual and real subjects from a HRTF database, predictions and the experimental evaluation aimed to assess the vertical localization performance with HRTF sets are selected by the proposed procedure. The results report a statistically significant improvement in predictions of the auditory model for localization performance with selected HRTFs compared to KEMAR HRTFs.
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Authors:Fenton, Steven; Lee, Hyunkook
Affiliation:Applied Psychoacoustics Laboratory (APL), The University of Huddersfield, Huddersfield, UK; Applied Psychoacoustics Laboratory (APL), The University of Huddersfield, Huddersfield, UK
This paper proposes and evaluates a perceptual model for the measurement of “punch” in musical signals based on a novel algorithm. Punch is an attribute that is often used to characterize music or sound sources that convey a sense of dynamic power or weight to the listener. A methodology is explored that combines signal separation, onset detection, and low-level feature measurement to produce a perceptually weighted punch score. The model weightings are derived through a series of listening tests using noise bursts, which reveal the perceptual relevance of the onset time and frequency components of the signal across octave bands. The punch score is determined by a weighted sum of these parameters using coefficients derived through regression analysis. The model outputs are evaluated against subjective scores obtained through a pairwise comparison listening test using a wide variety of musical stimuli and against other computational models. The model output PM95 outperformed the other models showing a “very strong” correlation with punch perception.
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Authors:Kahles, Julen; Esqueda, Fabián; Välimäki, Vesa
Affiliation:Aalto University, Acoustics Lab, Dept. of Signal Processing and Acoustics, Espoo, Finland; Aalto University, Acoustics Lab, Dept. of Signal Processing and Acoustics, Espoo, Finland; Aalto University, Acoustics Lab, Dept. of Signal Processing and Acoustics, Espoo, Finland
An assessment of filters for classic oversampled audio waveshaping schemes is carried out, pursuing aliasing reduction. For this purpose, the quality measure of the A-weighted noise-to-mask ratio is computed for test tones covering the frequency range from 27.5 Hz to 4.186 kHz, sampled at 44.1 kHz, and processed at eight-times oversampling. All filters are designed to have their passband contained within a ±1 dB range and to exhibit a minimum stopband attenuation value of 40 dB. Waveshaping of sinusoids via hard clipping is investigated—spectral enrichment due to the discontinuities introduced by its nonlinear transfer function maximizes aliasing distortion. The obtained results suggest that linear interpolation equalized with a high shelving filter is a sufficiently good method for upsampling. Concerning decimation, the interpolated FIR, elliptic, and cascaded integrator-comb filters all improve the results with respect to the trivial case. The cascaded integrator-comb filter is the only tested decimation filter that achieves perceptually sufficient aliasing suppression for the entire frequency range when combined with the linear interpolator.
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When attempting to synthesize virtual auditory environments the objects used for graphics and physics processing in games may not prove entirely satisfactory. There’s also the interesting possibility of capturing the boundaries and furniture within a space using 360° cameras, enabling room acoustic simulation. Reactive virtual acoustic environments can respond to dynamic directional cues from the sound source. Three possible application areas of IIA technology included the use of a rhythm-based game for stroke rehabilitation, a system for remote diagnosis of engine-room faults in ships, and an interactive spatial audio game to help children with autism.
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