Factors affecting the loudness of modulated sounds.

Loudness matches were obtained between unmodulated carriers and carriers that were amplitude modulated either periodically (rates between 2 and 32 Hz, modulation sinusoidal either on a linear amplitude scale or on a dB scale; the latter is called dB modulation) or with the envelope of the speech of a single talker. The carrier was a 4-kHz sinusoid, white noise, or speech-shaped noise. Both normally hearing subjects and subjects with cochlear hearing loss were tested. Results were expressed as the root-mean-square (rms) level of the modulated carrier minus the level of the unmodulated carrier at the point of equal loudness. If this difference is positive, this indicates that the modulated carrier has a higher rms level at the point of equal loudness. For normally hearing subjects, the results show: (1) For a 4000-Hz sinusoidal carrier, the difference was slightly positive (averaging about 0.7 dB). There was no significant effect of modulation rate or level over the range 20-80 dB SL. (2) For a speech-shaped noise or white noise carrier, the difference was close to zero, although for large modulation depths it tended to be negative. There was no clear effect of level (over the range 35-75 dB SPL) or modulation rate. For the hearing-impaired subjects, the differences were small, but tended to be slightly negative for both the 4000-Hz carrier and the noise carriers, when the modulation rate was above 2 Hz. Again, there was no clear effect of overall level. However, for dB modulation, the differences became more negative with increasing modulation depth. For modulation rates in the range 4-32 Hz, the results could be fitted reasonably well using the assumption that the loudness of modulated sounds is based on the rms value of the time-varying intensity of the response of the basilar membrane (taking into account the compression that occurs in the normal cochlea). The implications of the results for the fitting of multi-band compression hearing aids and for the design of loudness meters are discussed.     

Memory for the locations of environmental sounds

The accuracy with which a single source of sound can be localized has been examined in many studies, but very few studies have examined the ability of participants to determine the absolute locations of multiple sources of sound. The current study assessed participants' abilities to determine and remember the locations of up to six sources of environmental sound that were positioned at a range of azimuths and elevations in virtual auditory space. In experiment 1, a sequence of one to six sounds was presented one, three, or five times in each trial and the target sound was nominated following presentation of the last sequence. In experiment 2, memory load was held constant by nominating the target sound prior to a single sequence presentation. Localization accuracy was observed to decrease as the number of sounds was increased to three or more under the conditions of experiment 1, but not those of experiment 2. In experiment 1, localization was more accurate when sequences were presented more than once. Pronounced primacy and recency effects were observed for the six sound conditions in experiment 1. An analysis of errors for those conditions indicated that immediate temporal errors, but not immediate spatial errors, were over-represented.     

The loudness of sounds that increase and decrease continuously in level

A sound at a low level is heard as much softer after having decreased continuously from higher levels than if presented after a period of silence at that same low level. Canévet [Acustica 61, 256-264 (1986)] demonstrated this phenomenon for a tone that (1) decreased from 65 to 20 dB in 180 s; he also presented a tone that (2) increased from 20 dB, or (3) was presented as pairs of bursts at various levels in random order. Below about 40 dB, loudness changed most rapidly in the decreasing condition so that, at 20 dB, the tone was judged ten times softer than in conditions (2) and (3). In the present experiments, magnitude estimation was used to examine the possible role of judgmental biases and adaptation in this rapid loudness decline, which we call decruitment. Results show that decruitment did not come about because subjects made many successive loudness judgments; loudness declined as much when a tone was judged only twice, at the beginning and end of its 180-s decrease. In contrast, interrupting the decreasing tone so that it was heard only at 70 dB and 160 s later at 30 dB greatly diminished the decruitment. Similarly, pairs of 500-ms tone bursts presented at successively lower levels instead of continously decreasing did not show decruitment, suggesting that sequential biases are irrelevant. The likely cause of decruitment is sensory adaptation.     

End level bias on direct loudness ratings of increasing sounds

Three experiments on loudness of sounds with linearly increasing levels were performed: global loudness was measured using direct ratings, loudness change was measured using direct and indirect estimations. Results revealed differences between direct and indirect estimations of loudness change, indicating that the underlying perceptual phenomena are not the same. The effect of ramp size is small for the former and important for the latter. A similar trend was revealed between global loudness and direct estimations of loudness change according to the end level, suggesting they may have been confounded. Measures provided by direct estimations of loudness change are more participant-dependent.     

Comparison of amplification systems in an auditorium

Intelligibility of speech at two positions in a large auditorium was compared for the public address system (PA) and two assistive listening systems: Frequency modulation of radio frequencies (FM) and modulation of infrared light waves (IR). Listening groups were: normal-hearing adults, hearing-impaired, hearing aid users, elderly, and non-native. Word-identification scores were obtained with the Modified Rhyme Tests. Analysis of variance indicated that the main effects of systems, groups, and listening position were significant. Also significant were the two-way interactions. For all groups, the assistive listening systems provided better scores than the PA system. The difference between the two systems was statistically significant, but very small. It can be concluded that both listening systems provide improved speech intelligibility for various types of listeners.     

Influence of impression of vehicle styling on loudness of acceleration sounds in cabin

The influence of participants' impressions of vehicle styling on the loudness of acceleration sounds was investigated. A series of images of luxury or sporty vehicles was presented to the participants as acceleration sounds were being replayed. The results indicated that participants who were frequent drivers felt that the sound associated with luxury vehicles was louder than that associated with sporty vehicles. However, participants who rarely drove perceived almost no difference between the loudness of the two vehicles types. Thus, the loudness was shown to depend on both the participants' impression of the vehicle and their driving frequency.     

厅堂中总声压级的修正计算

完全扩散声场中的总声压级计算公式是假设混响声能在厅堂中的分布各处相等，与接收点的位置无关，实测发现该公式的计算值偏高，特别当接收点在临界距离以外时，本文提出了一个总声压级计算的修正公式，它与实测及计算机模拟的值符合较好。     

Experimental examination of vortex-sound generation in an organ pipe: A proposal of jet vortex-layer formation model

Aero-dynamical models of sound generation in an organ pipe driven by a thin jet are investigated through an experimental examination of the vortex-sound theory. An important measurement requirement (acoustic cross-flow as an irrotational potential flow reciprocating sinusoidally) from the vortex-sound theory is carefully realized when the pipe is driven with low blowing pressures of about 60 Pa (jet velocities of about 10 m/s). Particle image velocimetry (PIV) is applied to measure the jet velocity and the acoustic cross-flow velocity over the mouth area at the same phase by quickly switching the jet drive and the loudspeaker-horn drive. The vorticity of the jet flow field and the associated acoustic generation term are evaluated from the measurement data. It is recognized that the model of the “jet vortex-layer formation” is more relevant to the sound production than the vortex-shedding model. The acoustic power is dominantly generated by the flow–acoustic interaction near the edge, where the acoustic cross-flow velocity takes larger magnitudes. The acoustic generation formula on the vortex sound cannot deny the conventional acoustical volume-flow model because of the in-phase relation satisfied between the acoustic pressure at the mouth and the acoustic volume flow into the pipe. The vortex layers formed along both sides of the jet act as the source of an accelerating force (through the “acceleration unbalance”) with periodically alternating direction to oscillate the jet flow and to reinforce the acoustic cross-flow at the pipe mouth.     

High-resolution plane-wave decomposition in an auditorium using a dual-radius scanning spherical microphone array

The spatial and temporal distribution of early reflections in an auditorium is considered important for sound perception. Previous studies presented measurement and analysis methods based on spherical microphone arrays and plane-wave decomposition that could provide information on the direction and time of arrival of early reflections. This paper presents recent results of room acoustics analysis based on a spherical microphone array, which employs high spherical harmonics order for improved spatial resolution, and a dual-radius spherical measurement array to avoid ill-conditioning at the null frequencies of the spherical Bessel function. Spatial-temporal analysis is performed to produce directional impulse responses, while analysis based on the windowed Fourier transform is employed to detect direction of arrival of individual reflections at selected frequencies. Experimental results of sound-field analysis in a real auditorium are also presented.     

Estimates of annoyance of sounds of different character

The character of a sound is defined as the weighted combination of all acoustic factors, not contained in $\text{L}{}_{\mathrm{<mtext>A</mtext>}}$, contributing to its annoyance.From this definition it follows that differences in annoyance due to sounds with equal $\text{L}{}_{\mathrm{<mtext>A</mtext>}}$ are differences in sound character. For the concept of sound character to have real significance it is necessary that listeners agree on the annoyance due to sounds with equal $\text{L}{}_{\mathrm{<mtext>A</mtext>}}$.This paper describes a listening experiment with a variety of sounds of equal $\text{L}{}_{\mathrm{<mtext>A</mtext>}}$. The annoyance due to the sounds was rated by twelve subjects. Their individual ratings show significant agreement. Moreover, their average rating correlates well (0·90) with the ratings obtained by Terhardt and Stoll in a similar experiment.     

厅堂音质中的响度评价

厅堂音质评价的各项指标中,响度是最重要和最基本的内容之一。但由于长期来缺乏合适的参量,因此迄今无法在完工后的厅堂中去测量这项指标,当然也难以在设计阶段对此参量进行估算.不少人常把仅仅适用于稳态声源和混响场的声场估算法（即以直达声加上混响声）作为厅堂内各处总声强的评价,无论从音质设计和现场测量来看,显然很不合适。近年Lehmann提出以声强指数G（Starkemass）（dB）作为评价参量是一个好的建议。但根据我们的研究结果来看,鉴于早期反射声对响度起主导作用,因此厅堂内各处的声强指数应取50ms（语言）和80ms（音乐）的早期反射声积分值更符合实际,。以代替t从0积分到∞的评价方法。因此G_50ms和G_80ms将分别用于评价厅堂内对语言和音乐的响度评价参量。     

Roughness of organ pipe sound due to frequency comb

In the sound spectrum of flue organ pipes in addition to the usual harmonic partials, sometimes a series of equidistant but not harmonic lines can be found. This phenomenon has been observed in the recorded sound of pipes from different pipe ranks. The second set of spectral lines is similar to "frequency combs" used in optics for accurate measurement of optical frequencies. Analysis of measured sound spectra with and without frequency comb and simulations are presented and discussed in the paper. The appearance of frequency combs in the sound spectrum is explained by a model that assumes the presence of a mouth tone in addition to the pipe sound. Mouth tone bursts are generated when the oscillating air jet passes the upper lip. The burst repetition frequency is locked to the fundamental frequency of the pipe and the bursts are coherent with a pulse-to-pulse phase shift. The phase shift explains the observed frequency offset of the frequency comb to the harmonic frequencies. The simulations also show that weak and fluctuating mouth tones cannot generate frequency comb due to a lack of coherence.     

Linking the sounds of dolphins to their locations and behavior using video and multichannel acoustic recordings

It is difficult to attribute underwater animal sounds to the individuals producing them. This paper presents a system developed to solve this problem for dolphins by linking acoustic locations of the sounds of captive bottlenose dolphins with an overhead video image. A time-delay beamforming algorithm localized dolphin sounds obtained from an array of hydrophones dispersed around a lagoon. The localized positions of vocalizing dolphins were projected onto video images. The performance of the system was measured for artificial calibration signals as well as for dolphin sounds. The performance of the system for calibration signals was analyzed in terms of acoustic localization error, video projection error, and combined acoustic localization and video error. The 95% confidence bounds for these were 1.5, 2.1, and 2.1 m, respectively. Performance of the system was analyzed for three types of dolphin sounds: echolocation clicks, whistles, and burst-pulsed sounds. The mean errors for these were 0.8, 1.3, and 1.3 m, respectively. The 95% confidence bound for all vocalizations was 2.8 m, roughly the length of an adult bottlenose dolphin. This system represents a significant advance for studying the function of vocalizations of marine animals in relation to their context, as the sounds can be identified to the vocalizing dolphin and linked to its concurrent behavior.     

上海大剧院观众厅的音质设计与研究

上海大剧院是目前我国规模最大、起点最高、技术最新、音质优良的现代化大剧院。本文重点介绍上海大剧院的规模特点、观众厅音质设计技术要求、体形设计特点、混响控制设计、声学模拟试验研究及音质性能和主观评价等。