Calibrating the Sabine and Eyring formulas.

Of the many available reverberation time prediction formulas, Sabine's and Eyring's equations are still widely used. The assumptions of homogeneity and isotropy of sound energy during the decay associated with those models are usually recognized as a reason for lack of agreement between predictions and measurements. At the same time, the inaccuracy in the estimation of the sound-absorption coefficient adds to the uncertainty of calculations. This paper shows that the error of incorrectly assumed sound absorption is more detrimental to the prediction precision than the inherent error in the formulas themselves. The proposed absorption calibration procedure reduces the differences between the measured and predicted reverberation time values, showing that an accuracy within ±10% from the target reverberation time values can be achieved regardless of the absorption distribution in a room. The paper also discusses the oft neglected air absorption of sound, which may introduce considerable bias to the measurement results. The need for an air-absorption compensation procedure is highlighted, and a method for the estimation of its parameters in octave bands is proposed and compared with other approaches. The results of this study provide justification for the use of the Sabine and Eyring formulas for reverberation time predictions.

Robust selection of clean swept-sine measurements in non-stationary noise.

The exponential sine sweep is a commonly used excitation signal in acoustic measurements, which, however, is susceptible to non-stationary noise. This paper shows how to detect contaminated sweep signals and select clean ones based on a procedure called the rule of two, which analyzes repeated sweep measurements. A high correlation between a pair of signals indicates that they are devoid of non-stationary noise. The detection threshold for the correlation is determined based on the energy of background noise and time variance. Not being disturbed by non-stationary events, a median-based method is suggested for reliable background noise energy estimation. The proposed method is shown to detect reliably 95% of impulsive noises and 75% of dropouts in the synthesized sweeps. Tested on a large set of measurements and compared with a previous method, the proposed method is shown to be more robust in detecting various non-stationary disturbances, improving the detection rate by 30 percentage points. The rule-of-two procedure increases the robustness of practical acoustic and audio measurements.

Perceptual analysis of directional late reverberation.

The late reverberation characteristics of a sound field are often assumed to be perceptually isotropic, meaning that the decay of energy is perceived as equivalent in every direction. In this paper, we employ Ambisonics reproduction methods to reassess how a decaying sound field is analyzed and characterized and our capacity to hear directional characteristics within late reverberation. We propose the use of objective measures to assess the anisotropy characteristics of a decaying sound field. The energy-decay deviation is defined as the difference of the direction-dependent decay from the average decay. A perceptual study demonstrates a positive link between the range of these energy deviations and their audibility. These results suggest that accurate sound reproduction should account for directional properties throughout the decay.

Audibility of aliasing distortion in sawtooth signals and its implications for oscillator algorithm design.

This paper investigates the audibility threshold of aliasing in computer-generated sawtooth signals. Listening tests were conducted to find out how much the aliased frequency components below and above the fundamental must be attenuated for them to be inaudible. The tested tones comprised the fundamental frequencies 415, 932, 1480, 2093, 3136, and 3951 Hz, presented at 60-dB SPL and 44.1-kHz sampling rate. The results indicate that above the fundamental the aliased components must be attenuated 0, 19, 26, 27, 32, and 41 dB for the corresponding fundamental frequencies, and below the fundamental the attenuation of 0, 3, 6, 11, 12, and 11 dB, respectively, is sufficient. The results imply that the frequency-masking phenomenon affects the perception of aliasing and that the masking effect is more prominent above the fundamental than below it. The A-weighted noise-to-mask ratio is proposed as a suitable quality measure for sawtooth signals containing aliasing. It was shown that the bandlimited impulse train, the differentiated parabolic waveform, and the fourth-order polynomial bandlimited step function synthesis algorithms are perceptually alias-free up to 1, 2, and 4 kHz, respectively. General design rules for antialiasing sawtooth oscillators are derived based on the results and on knowledge of level-dependence of masking.

Perceptually informed synthesis of bandlimited classical waveforms using integrated polynomial interpolation.

Digital subtractive synthesis is a popular music synthesis method, which requires oscillators that are aliasing-free in a perceptual sense. It is a research challenge to find computationally efficient waveform generation algorithms that produce similar-sounding signals to analog music synthesizers but which are free from audible aliasing. A technique for approximately bandlimited waveform generation is considered that is based on a polynomial correction function, which is defined as the difference of a non-bandlimited step function and a polynomial approximation of the ideal bandlimited step function. It is shown that the ideal bandlimited step function is equivalent to the sine integral, and that integrated polynomial interpolation methods can successfully approximate it. Integrated Lagrange interpolation and B-spline basis functions are considered for polynomial approximation. The polynomial correction function can be added onto samples around each discontinuity in a non-bandlimited waveform to suppress aliasing. Comparison against previously known methods shows that the proposed technique yields the best tradeoff between computational cost and sound quality. The superior method amongst those considered in this study is the integrated third-order B-spline correction function, which offers perceptually aliasing-free sawtooth emulation up to the fundamental frequency of 7.8 kHz at the sample rate of 44.1 kHz.

Method for measuring violin sound radiation based on bowed glissandi and its application to sound synthesis.

This work presents a method for measuring and computing violin-body directional frequency responses, which are used for violin sound synthesis. The approach is based on a frame-weighted deconvolution of excitation and response signals. The excitation, consisting of bowed glissandi, is measured with piezoelectric transducers built into the bridge. Radiation responses are recorded in an anechoic chamber with multiple microphones placed at different angles around the violin. The proposed deconvolution algorithm computes impulse responses that, when convolved with any source signal (captured with the same transducer), produce a highly realistic violin sound very similar to that of a microphone recording. The use of motion sensors allows for tracking violin movements. Combining this information with the directional responses and using a dynamic convolution algorithm, helps to improve the listening experience by incorporating the violinist motion effect in stereo.

Analysis of the part-pedaling effect in the piano.

This letter reports basic acoustic phenomena related to part-pedaling in the piano. With part-pedaling, the piano tone can be divided into three distinct time intervals: initial free vibration, damper-string interaction, and final free vibration. Varying the distance of the damper from the string, the acoustic signal and the damper acceleration were measured for several piano tones. During the damper-string interaction, the piano tone decay is rapid and the timbre of the tone is affected by the nonlinear amplitude limitation of the string motion. During the final free decay, the string continues to vibrate freely with a lower decay rate.

The role of musical aptitude and language skills in preattentive duration processing in school-aged children.

We examined 10-12-year old elementary school children's ability to preattentively process sound durations in music and speech stimuli. In total, 40 children had either advanced foreign language production skills and higher musical aptitude or less advanced results in both musicality and linguistic tests. Event-related potential (ERP) recordings of the mismatch negativity (MMN) show that the duration changes in musical sounds are more prominently and accurately processed than changes in speech sounds. Moreover, children with advanced pronunciation and musicality skills displayed enhanced MMNs to duration changes in both speech and musical sounds. Thus, our study provides further evidence for the claim that musical aptitude and linguistic skills are interconnected and the musical features of the stimuli could have a preponderant role in preattentive duration processing.

Musical aptitude and second language pronunciation skills in school-aged children: neural and behavioral evidence.

The main focus of this study was to examine the relationship between musical aptitude and second language pronunciation skills. We investigated whether children with superior performance in foreign language production represent musical sound features more readily in the preattentive level of neural processing compared with children with less-advanced production skills. Sound processing accuracy was examined in elementary school children by means of event-related potential (ERP) recordings and behavioral measures. Children with good linguistic skills had better musical skills as measured by the Seashore musicality test than children with less accurate linguistic skills. The ERP data accompany the results of the behavioral tests: children with good linguistic skills showed more pronounced sound-change evoked activation with the music stimuli than children with less accurate linguistic skills. Taken together, the results imply that musical and linguistic skills could partly be based on shared neural mechanisms.

Analysis and modeling of piano sustain-pedal effects.

This paper describes the main features of the sustain-pedal effect in the piano through signal analysis and presents an algorithm for simulating the effect. The sustain pedal is found to increase the decay time of partials in the middle range of the keyboard, but this effect is not observed in the case of the bass and treble tones. The amplitude beating characteristics of piano tones are measured with and without the sustain pedal engaged, and amplitude envelopes of partial overtone decay are estimated and displayed. It is found that the usage of the sustain pedal introduces interesting distortions of the two-stage decay. The string register response was investigated by removing partials from recorded tones; it was observed that as the string register is free to vibrate, the amount of sympathetic vibrations is increased. The synthesis algorithm, which simulates the string register, is based on 12 string models that correspond to the lowest tones of the piano. The algorithm has been tested with recorded piano tones without the sustain pedal. The objective and subjective results show that the algorithm is able to approximately reproduce the main features of the sustain-pedal effect.

Fast automatic inharmonicity estimation algorithm.

A new algorithm is presented for estimating the inharmonicity coefficient of slightly inharmonic stringed instrument sounds. In the proposed partial frequencies deviation method, the inharmonicity is estimated in an intuitive way by minimizing the deviation of the expected partial frequencies compared to the frequencies of the high amplitude peaks in the spectrum. This is done in an iterative process, where the algorithm converges towards the target estimation value. The algorithm is tested using both synthetic and recorded piano tones. The results show that the new algorithm produces accurate results with a small computation cost compared to other methods.

Model-based sound synthesis of the guqin.

This paper presents a model-based sound synthesis algorithm for the Chinese plucked string instrument called the guqin. The instrument is fretless, which enables smooth pitch glides from one note to another. A version of the digital waveguide synthesis approach is used, where the string length is time-varying and its energy is scaled properly. A body model filter is placed in cascade with the string model. Flageolet tones are synthesized with the so-called ripple filter structure, which is an FIR comb filter in the delay line of a digital waveguide model. In addition, signal analysis of recorded guqin tones is presented. Friction noise produced by gliding the finger across the soundboard has a harmonic structure and is proportional to the gliding speed. For pressed tones, one end of a vibrating string is terminated either by the nail of the thumb or a fingertip. The tones terminated with a fingertip decay faster than those terminated with a thumb. Guqin tones are slightly inharmonic and they exhibit phantom partials. The synthesis model takes into account these characteristic features of the instrument and is able to reproduce them. The synthesis model will be used for rule based synthesis of guqin music.

Cortical correlates of acquired deafness to dissonance.

Patient I.R., who had bilateral lesions in the auditory cortex but intact hearing, did not distinguish dissonant from consonant musical excerpts in behavioral testing. We additionally found that the electrical brain responses did not differentiate musical intervals in terms of their dissonance/consonance, consistent with the idea that this phenomenon depends on the integrity of cortical functions.

Acoustical analysis and model-based sound synthesis of the kantele.

The five-string Finnish kantele is a traditional folk music instrument that has unique structural features, resulting in a sound of bright and reverberant timbre. This article presents an analysis of the sound generation principles in the kantele, based on measurements and analytical formulation. The most characteristic features of the unique timbre are caused by the bridgeless string termination around a tuning pin at one end and the knotted termination around a supporting bar at the other end. These result in prominent second-order nonlinearity and strong beating of harmonics, respectively. A computational model of the instrument is also formulated and the algorithm is made efficient for real-time synthesis to simulate these features of the instrument timbre.