Cats exhibit the Franssen Effect illusion

The Franssen Effect (FE) is a striking auditory illusion previously demonstrated only in humans. To elicit the FE, subjects are presented with two spatially-separated sounds; one a transient tone with an abrupt onset and immediate ramped offset and the other a sustained tone of the same frequency with a ramped onset which remains on for several hundred ms. The FE illusion occurs when listeners localize the tones at the location of the transient signal, even though that sound has ended and the sustained one is still present. The FE illusion occurs most readily in reverberant environments and with pure tones of approximately 1-2.5 kHz in humans, conditions where sound localization is difficult in humans. Here, we demonstrate this illusion in domestic cats using, for the first time, localization procedures. Previous studies in humans employed discrimination procedures, making it difficult to link the FE to sound localization mechanisms. The frequencies for eliciting the FE in cats were higher than in humans, corresponding to frequencies where cats have difficulty localizing pure tones. These findings strengthen the hypothesis that difficulty in accurately localizing sounds is the basis for the FE.     

Extension of a binaural cross-correlation model by contralateral inhibition. I. Simulation of lateralization for stationary signals

Running interaural cross correlation is a basic assumption to model the performance of the binaural auditory system. Although this concept is particularly suited to simulate psychoacoustic localization phenomena, there exist some localization effects which cannot be explained by pure cross correlation. In this paper a model of interaural cross correlation is extended by a "contralateral-inhibition mechanism" and by "monaural detectors" in order to simulate a wide range of psychoacoustic lateralization data. The extended model explains lateralization of pure tones with interaural time differences as well as with interaural level differences. Multiple images are predicted for tones with characteristic combinations of interaural signal parameters and for noise signals with different degrees of interaural cross correlation. The model is also capable of simulating dynamic lateralization phenomena, such as the "law of the first wave front" which is dealt with in a companion paper [Lindemann, J. Acoust. Soc. Am. 80, 1623-1630 (1986)]. The present paper is restricted to a comparison of the model predictions for stationary signals with the results of dichotic listening experiments.     

Phase effects on the perceived elevation of complex tones

Free-field source localization experiments with 30 source locations, symmetrically distributed in azimuth, elevation, and front-back location, were performed with periodic tones having different phase relationships among their components. Although the amplitude spectra were the same for these different kinds of stimuli, the tones with certain phase relationships were successfully localized while the tones with other phases led to large elevation errors and front-back reversals, normally growing with stimulus level. The results show that it is not enough to have a smooth, broadband, long-term signal spectrum for successful sagittal-plane localization. Instead, temporal factors are important. A model calculation investigates the idea that the tonotopic details that mediate localization need to be simultaneously, or almost simultaneously, accessible in the auditory system in order to achieve normal elevation perception. A qualitative model based on lateral inhibition seems capable in principle of accounting for both the phase effects and level effects.     

Underwater auditory localization by a swimming harbor seal (Phoca vitulina)

The underwater sound localization acuity of a swimming harbor seal (Phoca vitulina) was measured in the horizontal plane at 13 different positions. The stimulus was either a double sound (two 6-kHz pure tones lasting 0.5 s separated by an interval of 0.2 s) or a single continuous sound of 1.2 s. Testing was conducted in a 10-m-diam underwater half circle arena with hidden loudspeakers installed at the exterior perimeter. The animal was trained to swim along the diameter of the half circle and to change its course towards the sound source as soon as the signal was given. The seal indicated the sound source by touching its assumed position at the board of the half circle. The deviation of the seals choice from the actual sound source was measured by means of video analysis. In trials with the double sound the seal localized the sound sources with a mean deviation of 2.8 degrees and in trials with the single sound with a mean deviation of 4.5 degrees. In a second experiment minimum audible angles of the stationary animal were found to be 9.8 degrees in front and 9.7 degrees in the back of the seal's head.     

Attenuation of hearing protectors for pure-tones and 13-octave bands of noise

Eleven hearing protectors were tested according to the Argentinian Standard (IRAM 4060) with pure tones and random noise filtered in 1/3-octave bands as the test stimuli. Although the differences between the standard deviations are not statistically significant, there are significant differences between mean values for the two types of test signal. This indicates that a re-examination of the standard method is required.     

Influence of fundamental frequency and source elevation on the vertical localization of complex tones and complex tone pairs

This study investigates the vertical localization of single complex tones (monads) and simultaneous complex tone pairs (dyads), especially as it is affected by their fundamental frequency and source elevation. Two complex tone timbres are considered: one consisting of five low-order harmonics, and the other of all odd harmonics (a square wave). Sound sources were at -15, 0, 15, and 30 deg from the horizontal plane at ear height. For eight subjects, this source array was in the median plane, and for a further nine subjects, it was directly to the subject's left (lateral plane). The subjects localized the angle of the auditory image(s) of one or two complex tones around the vertical plane containing the sound sources. Mean responses for the five-harmonic complex tones show a systematic effect (referred to as Pratt's effect) of fundamental frequency on vertical localization--whereby high-frequency complex tones are localized to positions higher than low-frequency complex tones for equivalent source positions. For the square wave, the sound-source position dominates localization, although some effect of fundamental frequency is evident for median plane sources.     

Analysis of vocal pulses in a speech signal

An algorithm for estimating the vocal pulse positions and durations in an actual speech signal is described. Testing of the algorithm shows that it outperforms the best of the competitor algorithms in accuracy on the average by a factor of two. The algorithm is less sensitive to spectrum distortions in telephone channels, to various types of noise, and to instability in duration and amplitude of pulses produced by the voice source. The accuracy of the pulse position estimate is sufficient for a synchronous speech signal analysis, while the speed of signal processing makes the algorithm suitable for real-time operation.     

Generation and noise analysis of a wide-band optical -frequency comb based on recirculating frequency shifter

By optimizing the gain configuration and length of the loop, a 90-tone optical frequency comb （OFC） is successfully generated based on recirculating frequency shifter structure. The peak-to-peak power fluctuation of the 90-tone OFC is 4.26 dB and the tone-to-noise ratio is higher than 19.17 dB. To further analyze the noise accumulation feature of the tones when travelling around the loop, linewidth of the tones is measured by delayed self-heterodyne interferometer structure. The result shows the linewidth of the tones deteriorates little during the recirculating process, indicating that the generated OFC is an ideal multi-wavelength source for high-speed communication svstems.     

Localization of targets in CW-LFM radars with combined modulating signal

An improved algorithm of detecting multiple targets by cw radars with linear frequency modulation (LFM) is presented. A combined modulating signal consisting of successive LFM and pure Doppler periods has been used. Processing of pure Doppler periods does not require large computational resources and the obtained results are more accurate. This construction of algorithm allows reducing the probability of false alarm in the localization regime and simplifying the processing in the detection regime.     

用于铯芯片级原子钟的4.596 GHz射频源研制

芯片级原子钟主要包括射频模块、物理封装模块以及其他的外围控制模块。射频模块的设计关系到芯片级原子钟的短期稳定度,所以射频模块在芯片级原子钟的设计时是非常重要的一部分。本文利用数字锁相环技术实现频率为4.596 GHz的射频源,射频源由三部分组成,包括小数分频频率综合器、压控振荡器和环路滤波器。数字锁相环具有相位噪声低,频谱稳定度高等特点。此外,由于小数分频频率综合器是可编程的,可以通过配置N分频器与R分频器实现输出频率的快速扫描。与此同时,根据相关公式,可以计算出三阶无源环路滤波器的近似参数值,所设计的环路滤波器具有300 kHz的环路带宽以及55的相位裕度。最后,整个基于数字锁相环技术实现的射频源通过仿真、硬件实现以及测试。测试结果显示,射频源的相位噪声为-74.02 dBc/Hz@300 Hz,符合芯片级原子钟射频源的设计要求。     

运动声源快速定位的声达时差法

针对声达时差法只能用于非运动声源定位的问题,本文提出一种运动声源快速定位方法。该方法以声达时差为基本定位原理,基于声源计算位置对多普勒效应进行解耦并进行声信号多普勒效应修正,根据三角定位方法构建声传播空间矩阵,以声源位置偏差度为目标基于单纯形优化搜索算法进行声源位置快速逼近,实现了对匀速直线运动的单声源的定位追踪,提高定位实时性。该方法将声达时差法拓展到运动声源的定位,同时解决了消除多普勒效应带来的计算过程复杂、运算量大的问题,仅用4个传声器就可实现运动声源的快速定位,突破了传统运动声源识别中对大传声器阵列的依赖。仿真实验和实车运动声源识别实验结果证明了该方法的有效性,本研究为短时发声运动声源的识别提供了一种简便、高效的方法。      

Sound localization of stereo reproduction with parametric loudspeakers

A parametric loudspeaker radiates an audible signal by the interaction of the primary wave that is amplitude modulated and is known as a super-directivity loudspeaker. The parametric loudspeaker is one of the prominent applications of nonlinear acoustics. So far, the applications have been limited monaural reproduction sound system for public address in museum, station, street etc. In this paper, we investigated sound localization of stereo reproduction using two parametric loudspeakers in comparison with that using two ordinary dynamic loudspeakers. In subjective tests, the binaural information ILD (Interaural Level Difference) or ITD (Interaural Time Delay) was focused on. To investigate the characteristics of sound localization in a wide listening area, three typical listening positions were picked up. Signals were 500 Hz, 1 kHz, 2 kHz and 4 kHz pure tones and pink noise. The used parametric loudspeaker was an equilateral hexagon. The subjective test led to the results that when the parametric loudspeakers were used, the listeners at the three typical listening positions perceived the correct sound localization of not only pure tone but also pink noise and when the ordinary dynamic loudspeakers were used, except for the case of pure tone with ITD, the tendency was almost similar to those using the parametric loudspeakers. The second subjective tests were conducted in order to investigate in details the difference between parametric loudspeakers and ordinary dynamic loudspeakers by increasing the number of subjects. In the case of ITD and 500 Hz using the ordinary dynamic loudspeakers, three types of sound localization were categorized, in which the reversed type was major and the normal and the other types were minor. The ILDs which were measured with a dummy head and were calculated with several formulas were almost the same and indicated the reasons of the reversed typed sound localization and a serious influence of the crosstalk. It was found that in the case of pure tone with ITD, the contradiction between the binaural information ILD and ITD is remarkable, because the directivity of the ordinary dynamic loudspeakers was so dull that the crosstalk components had a serious influence on sound localization. It was determined the parametric loudspeaker could transmit correct binaural information to the listener, because the directivity of the parametric loudspeakers was so sharp that it suppressed the cross talk components.     

相干噪声快速去除算法及其在声源定位中的应用

针对相干噪声干扰声源辨识问题,将强跟踪滤波器理论与阵列-信号采集模型相结合,发展了一种快速估计相位变化的算法.算法引入多重次优渐消因子,能够进一步提取相位残差中的有用信息,使输出残差序列处处正交,且该因子能根据噪声相位差变化自动调节.通过仿真对连续相位突变进行跟踪表明,在参数失配条件下该算法实现了相位差的准确估计,且其...     

基于稀疏表示和特征加权的离格双耳声源定位*

基于头相关传递函数数据库的传统双耳声源定位方法的定位角度往往被限定在头相关传递函数数据库的离散测量点上。当头相关传递函数数据库的测量方位角间隔较大时,这类算法的性能会显著下降,这就是典型的离格问题。该文提出了基于加权宽带稀疏贝叶斯学习的离格双耳声源定位算法。首先该算法建立离格双耳信号的稀疏表示模型,然后利用双耳相干与扩散能量比特征对各个频点进行加权以降低噪声和混响的影响,最后通过加权宽带稀疏贝叶斯学习方法估计离格声源的方位角。实验结果表明,该算法在各种复杂的声学环境下都有着较高的定位精度和鲁棒性,特别是提高了离格条件下的声源定位性能。     

分布式压缩感知麦克风阵列多声源方位估计*

麦克风阵列已被广泛应用于音/视频会议等人机交互领域中时,多声源应用场景对声源方位估计性能提出了更高的要求。压缩感知(CS)声源定位算法将声源定位问题转化为信号的稀疏重构问题,相比传统的定位算法如相位变换加权（SRP-PHAT）和时延累加定位（DS）能够获得较高的定位性能,但多声源的存在一定程度上降低了稀疏程度,影响了CS重构性能。考虑到传统的CS定位算法并未利用多个连续语音帧之间声源空间向量的共同稀疏性,提出采用分布式压缩感知（DCS）理论以改善多声源的稀疏恢复估计的性能。仿真和实验结果表明,相比于传统定位算法和CS-OMP算法,DCS-SOMP算法在不同信噪比和不同声源强度的环境中,对多声源的方位估计都具有更好的定位性能和定位稳健性。     

A time-domain approach to estimating the plucking point of guitar tones obtained with an under-saddle pickup

A method for estimating the plucking point of guitar tones is proposed. The algorithm is based on investigating the time lag between two consecutive pulses arriving at the bridge of the guitar. The signal is detected with an under-saddle pickup attached to the bridge. The method determines the minimum of the autocorrelation function for one period of the signal. The time lag of the minimum can be converted into the distance from the bridge where the string was plucked. The results obtained with the method are good, the error remains smaller than one centimetre, except for a few outliers. The algorithm is easy to implement and can be used to analyse playing styles. The efficiency of the method gives the potential to also use it in real-time computer music applications.     

Decision rules in detection of simple and complex tones

Detection of simple and complex tones in the presence of a 64-dB SPL uniformly masking noise was examined in two experiments. In both experiments, the signals were either pure tones (220, 1100, or 3850 Hz) or an 18-tone complex consisting of equally intense components between 110 and 7260 Hz. In experiment 1, psychometric functions were obtained for detection in a 2I, 2AFC task. Results for eight normal listeners show that the psychometric functions are parallel for simple and complex tones. As expected, the masked thresholds for the pure tones are 43-44 dB SPL independent of frequency; the masked threshold for the complex tone is about 37 dB SPL per tone. These results indicate that the simultaneous presence of signal energy in many auditory channels aids detection. In experiment 2, psychometric functions were obtained with all four signals presented in random order within a block of trials. Results for four normal listeners show that the psychometric functions are parallel to one another and to those obtained in experiment 1. The thresholds are elevated to about 46 dB for the pure tones and to 40.5 dB for the complex tone. These results are nearly, but not quite, consistent with a multiband energy-detector model using an optimum decision rule; it appears that listeners may only make an unweighted sum of decision variables across an optimum selection of channels.     

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.     

Three-dimensional sound source localization using distributed microphone arrays

To improve the performance of sound source localization based on distributed microphone arrays in noisy and reverberant environments,a sound source localization method was proposed.This method exploited the inherent spatial sparsity to convert the localization problem into a sparse recovery problem based on the compressive sensing(CS) theory.In this method two-step discrete cosine transform(DCT)-based feature extraction was utilized to cover both short-time and long-time properties of the signal and reduce the dimensions of the sparse model.Moreover,an online dictionary learning(DL) method was used to dynamically adjust the dictionary for matching the changes of audio signals,and then the sparse solution could better represent location estimations.In addition,we proposed an improved approximate l_0norm minimization algorithm to enhance reconstruction performance for sparse signals in low signal-noise ratio(SNR).The effectiveness of the proposed scheme is demonstrated by simulation results where the locations of multiple sources can be obtained in the noisy and reverberant conditions.