中国筝的声功率级测试

筝是中国古老的弹弦乐器。但迄今为止,对其声功率级一直未进行科学的测定。本文在一混响室内根据ISO及GB标准,对二十一弦筝的声功率级进行了首次测定。两位资深乐师在混响室内分别演奏各自的乐器,通过围绕乐师和乐器布置的四通道测试设备,对筝所辐射的声功率级和动态范围进行测定。测试结果表明,中国筝在以不同力度演奏单音、音阶和乐曲时所辐射的声功率级及其频率特性均有所不同。考虑到乐器演奏音阶时所辐射的声功率级及其动态范围与演奏乐曲时的声功率级接近,并且,音乐的空间感也大都在乐器以f力度演奏乐曲的强音标志乐段时最为显著,故此我们建议中国筝所辐射的声功率级用其以f力度演奏音阶时的平均声功率级表示。本文测试的两架中国筝以f力度演奏音阶时的平均声功率级为85.9dB。文中不仅首次公布了中国筝声功率级的测试结果,并且所介绍的测试方法对其它乐器声功率级测试也具有借鉴意义。民族乐器所辐射的声音性能的确定是民族音乐厅堂音质研究的基础。     

瓦状阻尼橡胶块对高铁车轮减振降噪的影响分析

依据声学测试标准,为了评价某型高铁车轮在安装不同形式橡胶块装置后的减振降噪效果,在半消声室内基于B&K振动噪声测试分析系统,对裸轮和橡胶块车轮开展振动声辐射室内测试实验,并基于有限元方法对车轮模态进行了仿真分析。测试结果可知:相比裸轮,WA、WB车轮模态阻尼比显著增加,车轮的减振效果明显,其中WA车轮的减振效果略优于WB车轮。径向激励下,WA车轮声功率级降低了8 dB(A),WB车轮声功率级降低了5.5 dB(A);轴向激励下,WA车轮声功率级降低了8.2 dB(A),WB车轮声功率级降低了6.2 dB(A)。分析可知橡胶块装置能有效抑制车轮的滚动噪声和曲线啸叫,对车轮的减振降噪有积极作用。     

中国科学院声学计量测试站通过国家技术监督局计量标准考核和认证

中国科学院声学计量测试站于1988年12月16日至17日按“中华人民共和国计量法”的要求通过了国家技术监督局的计量标准考核和计量认证。通过考核的计量标准有:电声检定标准装置(一级)、水声换能器校准测试装置(一级)和压电加速度计检定标准(二级),由此该站可以进行声学量值的传递,检定标准电容传声器、声级校准器、声级计、标准噪声源、标准水听器及压电加速度计等声学计量器具和带通滤波器、声频放大器等声学测量仪器。通过计量认证的有关产品质量检验的测试项目有:声学实验室(消声室、半消声室、混响室等)性能测试、噪声源(机电产品)声功率级测试、环境噪声和工业噪声测试、厅堂音质测试、水     

线阵波束形成声强缩放方法估算声源的辐射声功率

为了解决波束形成声源识别过程中声源辐射声功率定量计算的问题,给出了阵型简洁、便于组合的线阵声强缩放模型。通过推导线阵的声强缩放系数,建立起线阵波束输出结果与声源辐射声功率之间的换算关系。无论是线阵还是平面阵的声强缩放方法,对于偏离阵列中心位置较远处的声源进行辐射声功率估算时都存在较为明显的误差。通过理论推导和仿真模拟计算,研究了同一单极子点声源在不同位置处的声功率估算偏差随频率、幅度的变化规律,发现该估算偏差只与声源偏离位置有关,而与声源自身的强度信息无关的结论,据此给出了相应的声功率估算修正方法。半消声室实验结果和声压法测量结果对比表明:修正后的线阵声强缩放方法用于中高频声源的辐射声功率计算时,单频声源的估算误差不超过1.0 dB,宽带声源的估算误差不超过1.8 dB。      

半消声室声场的研究

本文运用作者以前提出的消声室声场的计算公式,算出了点声源在半消声室内声场的解。其结果与声源离地面的高度有明显关系。从而定量地指出了鉴定半消声室时所知的声场与测试机器时的实际声场的区别,特别是纯青的声场,这种区别与人们以前的预想相差甚大。     

中国粤剧旦角发声特性实验研究

戏曲演员发声特性是厅堂音质研究的重要基础数据。为研究中国粤剧旦角演员发声特性,在全消声室内以4位花旦、2位青衣和1位老旦为例,针对念白、清唱和语言三种发声方式,在大、中、小三种发声力度下,分析不同旦角演员在0.3 m处的声压级、动态范围和频率特性曲线,探讨发声特性与发声方式、发声内容、发声力度和行当之间的关系。结果显示在相同的发声力度下,声压级由大到小的发声方式依次为清唱、念白和语言;老旦的动态范围最大,其次是青衣和花旦;清唱内容对发声频率特性曲线影响较小,而发声方式和行当影响较大;演员以不同力度发声时各频率特性曲线形状相似。该文的实验结果将为戏曲厅堂音质设计、建筑物隔声设计和电声系统设计提供科学依据。     

混响室内声源的声功率输出及声强分布

本文讨论了混响室内的声强分布,指出混响室内声强分布与自由场一样,对点声源服从平方反比律。对混响室及消声室的声压及声强随时间的起伏作了初步摸索,得到了几条实验规律,指出声强起伏比声压起伏更大。采用声强测量方法对同—声源在消声室及混响室内的声功率输出作了测量,说明声源的声功率输出是随环境变化的声学量,在混响室内声源的低频发射要比消声室内的发射要低。     

用简易声强计测量声强

本文介绍作者装置的一台简易声强计,实测误差与理论分析结果符合良好。组成声强计传感器的两个传声器之间的相对相移对测量结果有明显的影响,特别是在频率低端会产生很大的误差,作者通过使用倒向平均法降低了误差。消声室和混响室里的模拟实验结果证实了声强测量能可靠地在现场确定声源的声功率。     

汉语耳语标准频谱的测量与计算

提出了与GB7348-87《耳语标准频谱》不同的汉语耳语功率谱密度级随频率的变化关系。在消声室中测量以提高测量信噪比,使用实时分析仪测量单个人耳语发音的长期声压频谱,并且对每个人的长期声压频谱做自归一化,通过数学方法将多个样本"混录",计算出汉语耳语的功率谱密度级。汉语耳语标准频谱的测量和计算结果可为一切产生、传输、接收和处理汉语耳语信号的系统及电声器件的设计提供依据。      

非阻塞喷注噪声的A计权声功率的计算

喷注噪声的频谱是相似的。同一种介质的非阻塞喷注噪声的声功率谱随喷注直径的减小或喷注速度的增加而向高频移动,因而喷注噪声A计权声功率与喷注直径和喷注速度都有关系。本文使用-A声功率的连续计权函数和非阻塞喷注噪声的经验声功率谱函数计算了喷注噪声的A计权声功率与总声功率比值随喷注马赫数的增加而减小的关系。这个关系也可以用于估算因频谱的移动多喷口所获得的噪声降低。     

电影音乐录音中的电平控制

声音信号的重要计量值为 VU 值和峰值,它们分别直观反映声音强度及失真情况。为了透彻分析和阐述电影音乐节目制作系统的特性,该文通过实验的方法,测量了大量乐器在典型演奏法下的 VU值和峰值。进而通过实验数据的结果,做出 P/VU概率分布曲线。各音乐信号的 P/VU分布情况决定了系统的电平储备,掌握此结论非常有助于进行电影音乐节目制作过程中的电平控制。     

Optimizating the seats surrounding the platform in a recital hall when considering source directivity

The influence of source directivity can be significant for solo and chamber music performances, particularly for arena type halls that have improved visual intimacy for the audience. Computer simulation was performed to analyze the possibility of optimizing the acoustical qualities of a 4000-m³ recital hall with a significant portion of seats surrounding the platform when considering source directivity. The effects of surface treatments, length-to-width proportion, seating arrangement and platform location were evaluated. Most acoustical parameters derived from a baritone singer source were categorized as one factor that can be best represented by 2-kHz band early sound strength G_E. Regardless of room proportion, front-back 2-kHz band energy difference derived from optimized models can be controlled within 4 dB, a value much smaller than the difference in direct component.     

Orthogonal acoustical factors of a sound field in a bamboo forest.

To investigate the acoustical quality of a sound field in a bamboo forest, acoustical measurements were conducted to obtain orthogonal acoustical factors of the sound field. These results are compared with previous results for a sound field in an ordinary forest [H. Sakai, S. Sato, and Y. Ando, J. Acoust. Soc. Am. 104, 1491-1497 (1998)]. The IACC, which is defined as a maximum value of the normalized interaural cross-correlation function between signals at the ears, was 0.07 (4 kHz) and 0.16 (2 kHz) at positions 20 and 40 m from the source, respectively. These values are much better than those in the previously investigated forest. The measured subsequent reverberation time Tsub was up to 1.5 s in the frequency range above 1 kHz at the position 40 m from the source. For certain music sources with higher frequency components, therefore, sound fields in a bamboo forest have excellent acoustic properties.     

Sound strength and reverberation time in small concert halls

Sound strength and reverberation time measurements have been carried out in six small concert halls in Cambridge, UK. The sound strength G is a measure of the physical sound level in a concert hall and is closely related to the subjective sensation of loudness. It compares integrated impulse responses at a point in the measured room with that measured at ten metres distance in the free field.The aim of the measurements is to investigate the acoustic characteristics of the halls concerning sound strength and reverberation time. Furthermore the effect of the variable acoustics in the halls on these parameters is discussed in this paper. Especially for bigger ensembles it is often desirable to reduce the sound level in a small concert hall. The measurement results show that for a fixed hall volume, this can primarily be achieved by decreasing the reverberation time in the hall. However, with regard to the sound quality of a hall and the recommended reverberation times for chamber music, reverberation time cannot be reduced by an arbitrary extent. Therefore reverberation time and strength have to be balanced very carefully in order to obtain sufficient reverberation whilst at the same time avoiding excessive loudness. Finally the measured strength levels are compared to values derived from traditional and revised theory [Barron M, Lee L-J. Energy relations in concert auditoriums. J Acoust Soc Am 1988;84(2):618-28] on strength calculations in order to assess the accuracy of the theory for small chamber music halls. Possible reasons for the low measured strength levels observed are discussed with reference to related design features and objective acoustic parameters.     

A scale value for the balance inside a historical opera house

In the framework of opera house acoustics, the term "balance" refers to the acoustical competition between the singer on the stage and the orchestra in the pit. The mechanism allowing the operatic singers to be heard over the orchestra has to do with their skill in enhancing the vocal emission by a peculiar use of the formant frequencies. This vital factor is sensed by the listeners and, apart from the obvious sound power ratio of the stage and the pit sources, is the main cue that helps to formulate a subjective impression of the balance. To achieve its objective qualification, two calibrated sound sources can be placed on the stage and in the pit, respectively, and their sound level difference is measured at the listeners' seats. The scope of this work is to investigate the relationship between the subjective impression and the objective indicator of the balance and to develop a scale value for the parameter in the case of a historical opera house. For this scope a set of acoustical data from the Teatro Comunale in Ferrara will be used to create synthetic sound fields with controlled conditions of the balance between the stage and the pit. This methodology employs an anechoic piece for soprano (with piano accompaniment) and is implemented in a dead room equipped with an acoustical rendering system. The sound fields are used to investigate the appropriate balance values by means of listening tests. The results of the scaling exercise show that a suitable range of values can be extracted and that the sound from the stage and the pit is perceived as balanced when the loudness difference between the two is comprised within -2.0 dBA and +2.3 dBA.     

Perceptual validation of virtual room acoustics: Sound localisation and speech understanding

The reliability of algorithms for room acoustic simulations has often been confirmed on the basis of the verification of predicted room acoustical parameters. This paper presents a complementary perceptual validation procedure consisting of two experiments, respectively dealing with speech intelligibility, and with sound source front–back localisation.The evaluated simulation algorithm, implemented in software ODEON®, is a hybrid method that is based on an image source algorithm for the prediction of early sound reflection and on ray-tracing for the later part, using a stochastic scattering process with secondary sources. The binaural room impulse response (BRIR) is calculated from a simulated room impulse response where information about the arriving time, intensity and spatial direction of each sound reflection is collected and convolved with a measured Head Related Transfer Function (HRTF). The listening stimuli for the speech intelligibility and localisation tests are auralised convolutions of anechoic sound samples with measured and simulated BRIRs.Perception tests were performed with human subjects in two acoustical environments, i.e. an anechoic and reverberant room, by presenting the stimuli to subjects in a natural way, and via headphones by using two non-individualized HRTFs (artificial head and hearing aids placed on the ears of the artificial head) of both a simulated and a real room.Very good correspondence is found between the results obtained with simulated and measured BRIRs, both for speech intelligibility in the presence of noise and for sound source localisation tests. In the anechoic room an increase in speech intelligibility is observed when noise and signal are presented from sources located at different angles. This improvement is not so evident in the reverberant room, with the sound sources at 1-m distance from the listener. Interestingly, the performance of people for front–back localisation is better in the reverberant room than in the anechoic room.The correlation between people’s ability for sound source localisation on one hand, and their ability for recognition of binaurally received speech in reverberation on the other hand, is found to be weak.     

Directivity pattern of neurosurgical endoscopic ultrasonic probes

The objective of this paper is to present recent investigations in characteristics of the sound field generated by neurosurgical endoscopic ultrasonic probes (NEUPs) for minimally invasive surgery. The importance of this information has been investigated and discussed taking into account following facts: 1. According to the International Standard IEC 61847:1998 basic acoustical output characteristics of ultrasonic surgical equipment is declared for and measured in an acoustical free field. The standard generally treats the ultrasonic probe as an omnidirectional point source of the zero order (monopole source). 2. In real conditions, operations with NEUPs are performed within the acoustical near field. Having in mind that the cavitational and hydrodynamic effects are dominant, two theoretical boundary conditions can be present. The first one takes place when operations are performed near the "soft" acoustical boundary (tissue/air), and the second one is near the "rigid" acoustical boundary (tissue/bone). Reflections of sound waves from boundaries have influence on the characteristics of the ultrasonic probe (transducer) and on the sound field. In such cases spherical waves of the first and second order are generated. Directivity of sound sources takes shape in the far field and is easier to measure there. On the basis of measured directivity patterns, the influence of different operational conditions (immersion depth of the probe tip, boundary type, acoustical impedance of the medium etc.) on the radiated sound power and spatial distribution of the sound pressure can be estimated.     

Saxophonists tune vocal tract resonances in advanced performance techniques

The acoustical impedance spectrum was measured in the mouths of saxophonists while they played. During bugling and while playing in the very high or altissimo range, experienced players tune a strong, but relatively broad, peak in the tract impedance to select which peak in the bore impedance will determine the note. Less experienced players are unable to produce resonances with impedance peaks comparable in magnitude to those of the bore and consequently are unable to play these notes. Experienced players can also tune their tracts to select which combinations of notes are played simultaneously in multiphonics or chords, and to produce pitch bending, a technique in which notes are produced at frequencies far from those of the peak of impedance of the instrument bore. However, in normal playing in the standard range, there is no consistent tuning of the tract resonances. The playing frequency, in all cases, lies close to the peak in the impedance of the reed in parallel with the series combination of the impedances measured in the mouth and the instrument bore on either side of the reed (ZMouth+ZBore)∥ZReed.