bluuebunny/crossref_metadata_2025 huggingface dataset:
Impact of MP3-compression on timbre space of sustained musical instrument tones
Authors: Lee Chung , Horner Andrew , Beauchamp James
Published: April 2012

MP3 compression is widely used in music sharing and storage. A number of studies have investigated the discrimination of instrument tones after MP3 compression. Additionally, a number of previous studies have evaluated other data reduction methods including frequency modulation (FM) synthesis, wavetable synthesis, and principal component analysis (PCA). However, these studies have not considered the impact on timbre space after data reduction. In this study, listening test subjects were asked to rate the dissimilarity of all pairs of original instrument tones. The same process was done on MP3-compressed tones with various bit-rates. Correlation analysis was done on the dissimilarity data of the original and compressed tones to see if MP3 compression caused a significant impact on the perceptual distance between instrument pairs. The multidimensional scaling (MDS) solutions of the original and compressed tones were also compared to see if the timbre space was significantly altered after MP3 compression (e.g., would a clarinet sound more or less similar to an oboe after MP3 compression?) [This work was supported by RGC grants 613510 and 613111.]
Individual differences in the relative salience of percussive timbre dimensions
Authors: McKinnon-Bassett Mark , Martens William L.
Published: April 2012

A set of nine percussive timbres was generated by varying parameters of two resonant filters incorporated in a hybrid resynthesis of recorded drum sounds. Three values of damping for a lower-frequency resonance were factorially combined with three center-frequency values for a higher-frequency resonance associated with percussive attack transients. Two groups of listeners were asked to produce dissimilarity judgments for all pairwise comparisons of the nine sounds on a ten-point scale. The dissimilarity judgment data from two groups of subjects were combined to form a single dataset for submission to Individual Differences Scaling (INDSCAL) analysis. A common timbre space of just two dimensions was derived along with a subject space that revealed the different weights placed by each subject on each of those dimensions of the derived timbre space. Individual differences in the relative salience of these percussive timbre dimensions were related to the musical training of the listeners.
Challenges in the implementation of active noise control technologies
Authors: Qiu Xiaojun , Li Ningrong
Published: April 2012

A number of projects have been carried out in Nanjing University to implement active noise control technologies, which include active control of transformer noise, active sound barrier, active noise control in communication chassis, active noise control in natural ventilation windows, active control of large impulsive noise in headset and active noise control in a train compartment. Unlike our previous research, these projects are all funded by industries and the aim is not for doing academic research, but to make commercial prototypes. Various challenges in the implementation of active noise control technologies in these projects to make commercial products are reported and discussed, and main issues to make successful commercial active noise control products are pointed out.
On family of fractional lower order moment (FLOM)-based algorithms for active noise control of impulsive noise sources
Authors: Akhtar Muhammad , Mitsuhashi Wataru
Published: April 2012

Active noise control (ANC) is based on the principle of destructive interference of propagating acoustic waves; essentially a cancelling signal is generated and combined with the primary noise to achieve acoustic cancellation around location of the error microphone. In this paper, we consider a very challenging application of ANC for impulsive noise. The impulsive noise can be modeled using non-Gaussian stable process for which second order moment does not exist. The most famous filtered-x-LMS (FxLMS) algorithm for ANC systems, is based on minimization of the variance of the error signal, and therefore, becomes unstable for the impulsive noise. It has been shown that filtered-x least mean p-power (FxLMP) algorithm; based on minimizing the fractional lower order moment (FLOM) that does exist for stable distributions; gives robust performance for impulsive ANC. However, the convergence speed of the FxLMP algorithm is very slow. Recently; we have proposed various variants of FxLMP algorithm, so that an improved convergence and noise reduction performance is achieved. In this paper, we propose modifying and employing generalized normalized LMP algorithm (GNLMP) algorithm for ANC of impulsive noise. The computational complexity of proposed algorithm is comparable to the existing FLOM-based ANC algorithms. Extensive simulations are carried out, which demonstrate the effectiveness of proposed algorithm. We observe that, in comparison with the existing FLOM-based ANC algorithms, the proposed algorithm gives best performance for ANC of impulsive noise sources.
The design of a multi-channel active noise controller with ultra low latency
Authors: Chen Kai , Lu Jing , Qiu Xiaojun
Published: April 2012

For wide band noise control, the performance of active noise control systems depends largely on the latency of the controller. The latency of the controller is usually caused by the AD and DA converter, and the normally used Delta-Sigma audio codec is not suitable for a real time active noise control system because several milliseconds time delay of the codec will lead to non-causality of the whole control system. In this paper, an ultra low latency multi-channel audio input and output system is described. The data of the system can be interacted with a float-point DSP, where a high efficient multi-channel feed-forward control algorithm is embedded. The proposed controller also includes an ARM processor, which is in charge of the friendly user interface.
Novel application of PVDF film in active noise control through windows
Authors: Lane Jeremy , Pearse John , Gutschmidt Stefanie
Published: April 2012

PVDF film has been used widely used in active control solutions for noise and vibration. In this work, due to the transparent property of PVDF film and the proven possibility of transparent electrodes, the feasibility of PVDF film's use in the construction of a second source for active noise control (ANC) through windows is considered. Sound pressure level measurements are described to establish the feasibility of PVDF in this application. Different configurations, using glass and acrylic glass backing materials of varying thicknesses, of PVDF film speaker are reported and compared. Finally comments are made on relative performance and the overall likelihood of use in an ANC application for windows.
An investigation on passive-active absorption system in a water-filled impedance tube
Authors: Wang Xiaolin , Liu Bilong , Tian Jing
Published: April 2012

A hybrid passive-active sound absorption structure is experimentally investigated in a water-filled impedance tube. The surface impedance of the passive material is manipulated by active parts to match the impedance of medium, and consequently to improve sound absorption performance at low-frequency range. Parameters such as boundary, material thickness and position of secondary source are optimized by numerical analysis to achieve high absorption coefficient. Experimental results show that the developed hybrid structure has potential to improve sound absorption at low frequency range.
Soundscape case study: acoustics, ecology and its anthropological sense
Authors: Deng Zhiyong , Zhou Guowen , Hua Wei , Wang Yun , Zhang Jian
Published: April 2012

Base on six soundscape investigation case studies in some small urban or historical areas, included the South Putuo Temple of Xiamen city in 2006, the downtown of Kashkar city in 2006, the local worship music performing area of South Gaoluo village in 2007, the Nanxiangkou local music performing area in Hebei University of Technology Gymnasium of ShijiaZhuang city in 2007, the downtown riverside of Liuzhou city in 2010, the local worship and music performing areas of Qianjuntai village and Zhuanghu village of Beijing city in 2011, a certain relationship analysis between the acoustical parameters and its audience's behavior or subjective assessment is put forward in this paper. Due to the two-dimensional (Leq and the subjective assessments) fuzzy clustering or curve fitting, it shows that the sound ecology would have a critical connection to the anthropological sense, which means some keynotes sound and soundmarks must be keep constantly during the soundscape design or the urban development. Furthermore, a rough concept of sound history and the constant of keynotes and soundmarks in the view of soundscape are also discussed in this paper.
Soundscape variation in a historical city centre due to new traffic regulation
Authors: Maffei Luigi , Di Gabriele Maria , Aletta Francesco
Published: April 2012

In recent years the number of candidate historic city centers to be included in the World Heritage List is increasing. This inclusion must be supported by a Management Plan programming all intervention to be implemented for the preservation of the “outstanding universal value”. So far the Management Plans do not consider the preservation and valorization of the soundscape. Consequently, urban renewal processes are based on conservation and restoration of tangible cultural heritage, in order to increase touristic attraction and to improve the quality of life. All these efforts privilege visual perception and do not take in account the auditory perception. Soundscape of a site can be considered an intangible cultural heritage to be preserved and valorized as it constitutes a peculiar characteristic of the place. It makes the place recognizable and attractive. Recently the historic centre of Naples (Italy), as World Heritage Site, has been under renewal and for sustainable mobility the largest restricted traffic area (ZTL) in Europe has been introduced. The results of soundwalks carried out in the historic center of Naples before and after the implementation of ZTL are presented. The variations of acoustical and other environmental parameters influencing the subjective perception of environmental quality are analyzed.
Ultrasonic set up for the assessment of the stability of a cylinder inserted in a solid
Authors: Mathieu Vincent , Anagnostou Fani , Soffer Emmanuel , Haiat Guillaume
Published: April 2012

The study aims at proposing a new experimental ultrasonic methodology for the estimation of the stability of a cylinder inserted in a solid. Such a technology may have various fields of application: aeronautics, car industry, mechanics or also surgery. The present prototype is dedicated to the study of the stability of dental implants. Cylindrical titanium implants were inserted in four groups of rabbit femurs, each group corresponding to a controlled level of stability of the cylinders. The 10 MHz ultrasonic response of the cylinder is processed to derive quantitative indicators based on the temporal variation of the signal amplitude. Analysis of variance (ANOVA) (p < 10-5) tests revealed statistical distributions of indicators significantly correlated with the stability of the cylinders. A numerical finite-difference time-domain model was considered in order to understand the origin of the different echoes and the importance of lateral wave propagation was evidenced. The numerical model also enabled to estimate the sensitivity of the indicators to variations in the material properties of the materials in contact with the cylinders.
The study of dissipative nonlinearity in oil sand
Authors: Liu Jiehui , Zhu Jinlin , Liu Xiaozhou , Gong Xiufen , Zhou Dao
Published: April 2012

The acoustic waves propagating in sand have the nonlinear dissipative phenomenon that the dissipative coefficient of acoustic waves with larger amplitude is smaller than that with smaller amplitude. The results of experimental investigations on the propagation of acoustic waves in oil sand with different oil content are presented in the paper. The nonlinear dissipative phenomenon in oil sand is studied and the analytical description is given to explain the phenomenon. It is found that the relative growth coefficient and the dissipative index are dependent on the oil content in oil sand. According to the dependence relationships between the sensitive coefficients and the oil content, a new prospective approach to measure the oil content in oil sand is provided in the paper for oil exploration.
A pipe-like one-way structure of acoustic wave
Authors: Yuan Bo
Published: April 2012

We proposed a pipe-like structure with linear materials to obtain the acoustic unidirectional transmission. The system consists of a bending pipe and a phononic crystal and designed by extending the idea of frequency selection in nonlinear acoustic diode into the mode selection of a linear acoustic system. The system has a significant transmission efficiency and good rectifying ratio which is designed to work in the air. We also experimentally realized the unidirectional transmission behavior for acoustic waves and the experimental results agree well with the theoretical simulation. This device is expected to have potential applications in ultrasonic devices such as acoustic diodes.
The surface acoustic waves in the phononic crystal surface based on the locally resonance mechanism
Authors: Li Yong
Published: April 2012

We proposed a two dimensional structure consisted of a resonant unit stubbed periodically on a phononic crystal plate with finite thickness to investigate the surface acoustic waves (SAWs). Numerical results shown that two types of SAWs, one is the classical Rayleigh SAWs, whereas the other is the scattering SAWs overcoming the limitation that SAWs can only exit below the bulk cone of the substrate, can be found in the structure. Furthermore, the band gaps of the SAWs are obtained, as well as slow modes of the SAWs are also observed in the band gaps of the phononic crystal substrate. It could open the probability to control effectively the propagation of these SAWs. The results should have the impact on the SAWs communications.
Utilizing negative diffraction effect in phononic crystal to realize resting X-shaped wave localization
Authors: Kan Weiwei , Cheng Jianchun
Published: April 2012

Motivated by the promising impacts in acoustic devices and applications that can efficiently trap, guide, and manipulate sonic waves, researches of Photonic Crystals (PCs) have received great attention in the past decade. Many anomalous refractive and diffractive effects, such as superprism effects and negative refraction, have been found in such PCs. In this report, by studying the wave packet superposed by Bloch modes in the isofrequency surface at unique points of the PC band structure, the negative diffraction effect in certain directions of the PCs is exploited to realize X-shaped wave localization settled in one point, which cannot exist at rest in homogeneous media.
Cortical representations of pitch: theories and experiments
Authors: Wang Xiaoqin
Published: April 2012

Pitch perception is one of the most important auditory perceptual phenomena. Its underlying neural mechanisms have not been well understood. Recent human imaging studies and neurophysiology experiments in non-human primate have begun to reveal possible neural coding mechanisms in the cerebral cortex. These studies have pointed to a specialized area in the rostral region of primate auditory cortex where harmonic pitch is extracted. How pitch-selective neurons in this cortical area extract harmonic pitch at the cellular level, however, is yet known. Moreover, it remained to be explored whether other auditory cortical areas process aspects of pitch that are not processed by this rostral pitch-region. An important issue in the study of cortical representations of pitch is whether pitch embedded in harmonic complex sounds is extracted and uniquely represented by a specific cortical area or a subset of neurons in that area. Simply showing that pitch information exists in neural firing in a cortical area is not an adequate demonstration of pitch processing mechanisms. (Research supported by NIH grant R01-DC003180)
Rapid plasticity in auditory and prefrontal cortex during active listening
Authors: Shamma Shihab
Published: April 2012

Humans and other animals often attend to sounds in their environment so as to approach mates and competitors, or to avoid predators. Numerous neural processes orchestrate the performance of these behavioral tasks, including sensory adaptive responses in the auditory cortex and executive control functions in the prefrontal cortex. The multitude of mechanisms observed to be involved in neurophysiological recordings from several auditory and prefrontal cortical fields in behaving ferrets will be reviewed. The findings reveal that rapid changes in auditory receptive fields take place only during task performance, and that these serve to enhance discrimination and detection of target stimuli from their backgrounds. Interestingly, the changes also depend on the meaning of the sounds (aversive or appetitive) and the level of behavioral performance.
State-dependent changes in background discharge of auditory core neurons in freely moving guinea pigs
Authors: Ojima Hisayuki , Taira Masato
Published: April 2012

In natural situations, sounds, such as predator noises and nursing and mating calls, are used as signals in determining an animal's behavior. Shifts in behavior are adaptively controlled by activation of higher-order brain regions. Utilizing forced change in behavior as a trigger, we show that brain state shifts affect the background discharge pattern of neurons in primary auditory cortex (A1). Single unit recordings were made from freely moving guinea pigs, which were impelled to shift from a passive/stationary state to an active/exploratory state by a sudden change in ambient illumination from light to dark. Upon the illumination shift, background discharge was reduced significantly for several minutes. This behavioral shift was induced even when the animals were actively engaged in eating food. Acoustic stimulation during the time of reduced background discharge resulted in an improved S/N ratio of the neuron's response. These neurons were localized almost exclusively in upper layer 5, from which cortical feedback to subcortical stations originate. In naturalistic situations, a subset of A1 neurons may transfer information about the cortical state to subcortical auditory stations. Supported by KAKENHI to H.O. (No. 22500368).
The whisper sensitive scale in the application of speaker identification
Authors: Lin Wei
Published: April 2012

In this paper, the frequency characteristics of numerical whispered speech were investigated by a filter bank analysis. It was shown that the first and the third formants were more important than the other formants in the speaker identification of Chinese whispered speech. The experiment showed that the 800-1200 Hz and 2800-3200 Hz ranges were the most significant frequency ranges in discriminating the speaker. Based on this result£, a new feature scale named whisper sensitive scale (WSS) was proposed to replace the common scale, the Me1 scale£, and to extract the cepstral coefficient from whispered speech signal. Furthermore, a speaker identification system in whispered speech was presented based on WSCC (the whisper sensitive cepstral coefficient. And the new system performed better in solving the problem of speaker identification of numerical whispered speech than the traditional method.
A modified weighted overlap and add-based spectral subtraction method
Authors: Sun Yang , Yuan Meng , Feng Haihong
Published: April 2012

The Weighted Overlap and Add (WOLA) method is an optimization of frequency modulated DFT filter banks. However, the decrement in frequency resolution of speech signal caused by WOLA leads to noise estimation error. In this study, a speech power compression and gain compensation method was proposed to solve the low-resolution problem from the WOLA process. Finally, a multi-band spectral subtraction method was developed by combining the WOLA and the spectral subtraction algorithm in frequency domain. The noise reduction degrees were determined by the individual Signal-to-Noise Ratios (SNRs) in each frequency bands. Objective evaluation of the proposed noise reduction method was performed based on Itakura-Saito Distortion Measure under different types of noise. Statistical results showed that this proposed method could improve noisy speech by 5 dB SNR with little distortion. This noise reduction method has an advantage on low computation load, real-time processing and good performance on noise reduction. So this method can be implemented in embedded devices, e.g. hearing aid and cochlear implants. # This work is supported by National Natural Science Foundation of China (11104316). and Shanghai Natural Science Foundation (11ZR1446000).
The target tracking based on cubature Kalman filter
Authors: Fang Yuanyuan
Published: April 2012

A new extension of Kalman Filter to nonlinear system-- Cubature Kalman Filter is introduced.This algorithm has its theory basis consisted of Gaussian Bayesian theory and Spherical-radial rules.In the light of the unique properties of Cubature Kalman Filter:a derivative-free on-line sequential-state estimator, computational complexity grows as n3 and it eases the curse-of-dimensionality problem.The paper focuses on studying the CKF algorithm in depth, and applys it to bearing-only maneuvering target tracking.Two typical tracking model about tracking maneuvering ship from both non-maneuvering and maneuvering platform is selected.The Monte-Carlo simulations' results illustrate that the CKF algorithm outperforms EKF in accuracy and calculation efficiency of filter, and which make Cubature Kalman Filter easy and feasible to broad application prospect.
Combining Capon and Bartlett spectral estimators for detection of multiple sinusoids in colored noise environments
Authors: Zheng Chengshi , Li Xiaodong
Published: April 2012

Detection of multiple sinusoids in colored noise environments has many potential applications, such as sonar, radar, and communication. Most of conventional algorithms often use the local signal-to-noise-ratio (LSNR) as a test statistic to detect the sinusoids, where the LSNR is estimated in the frequency domain by using the Bartlett spectral estimator (BSE). Unfortunately, the BSE has a relatively low frequency resolution, which may degrade the detection performance significantly. To solve the frequency resolution problem of the BSE, this paper proposes a two-stage hybrid algorithm to estimate the LSNR. In the first stage, the BSE is used to estimate the noise power spectral density over frequency. After obtaining the noise power spectral density, the second stage employs the Capon spectral estimator (CSE) to estimate the LSNR. The proposed hybrid algorithm significantly improves the detection performance, especially when the sinusoids are closely spaced. Simulation results show that the proposed algorithm performs much better than the conventional algorithms in most cases.
A spatial domain processing method for the direct signal separation in the reverberant field
Authors: Wu Benyu
Published: April 2012

A spatial domain processing method for the direct signal separation from the same frequency reflections at different frequencies in the reverberant field is proposed. The reflections with the same frequency contained in the reverberant acoustic signals are distributed uniformly and irregularly in space. With the same direct signal being kept, the method makes the reverberant acoustic signals in the different spatial positions synchronized and superposed. Simultaneously the same frequency reflections are eliminated, and then the direct signal is separated. It overcomes the difficulty that the direct signal can't be obtained because of the effect of reflections. It is important and significant for the acoustic measurement or the other researches based on the direct signal. The results show that the method is verified feasible and effective, by the experiments at different frequencies.
Interoperability of heterogeneous cores on acoustic signal processing and communication system
Authors: Cao Shengchen , Yan Zhaoli , Cui Tianhao , Cheng Xiaobin , Li Xiaodong
Published: April 2012

An acoustic signal processing and communication system is typically constituted with FPGA, DSP and application processors. These heterogeneous cores have different speed, throughput and peripheral interfaces, which results in barriers of data sharing among them. Texas Instruments' open-source component DSPLINK is an applicable solution for ARM and DSP communication. In this paper, DSPLINK architecture is optimized, FPGA is designed to act as part of the memory pool and SPI based synchronization protocol between the three cores is implemented. As an example of application, a system for cavitations monitoring based on OMAP and FPGA platform is introduced subsequently. Linux OS and DSP/BIOS are ported to ARM end and DSP end respectively, based on which the interoperable architecture is implemented. The developers can use interoperable APIs for data transfer so that more applications can be easily derived.
The realization of precision time protocol for distributed acoustic and vibration measurements
Authors: Ma Longhua
Published: April 2012

Distributed acoustic and vibration measurement based on Local Area Network (LAN) becomes a hot topic, recently. The performance of this measurement system is affected dramatically by the synchronization precision of different measurement nodes. Traditional synchronous methods, such as Network Time Protocol (NTP), Simple Network Time Protocol (SNTP) doesn't meet the precision needed for distributed acoustic and vibration measurements, because it can only achieves accuracy of microseconds. In addition, sync cable is not suitable for long distance distributed acoustic and vibration measurement due to the inconsistent delay. To sovle this problem, a method based on Precision Time Protocol (PTP, IEEE1588) is proposed in this paper for synchronization of distributed acoustic and vibration measurement device. A Field Programmable Gate Array (FPGA) is employed between Medium Independent Interface (MII) and PHY, which monitors all ingress and egress data packets. Only PTP packets are unpacked and proceeded. According to these packets, the time offsets are calculated. Then time offsets between master clock and slave clock are filtered, and the output of the filter is used to compensate the drift of crystal oscillator. Synchronization experiment show the proposed method can achieve synchronization accuracy of few hundred nanoseconds.
Harmonic distortion measurement for a parametric loudspeaker with logarithmic time stretched pulse
Authors: Masunaga Shohei , Ikefuji Daisuke , Morise Masanori , Nishiura Takanobu
Published: April 2012

A parametric loudspeaker which utilizes the ultrasound can transmit the audible sound to a particular area. However, the sound reproduced by a parametric loudspeaker contains the harmonic distortions because the sound is demodulated by the nonlinearity in the air. Thus, measuring the harmonic distortions is required to evaluate the sound quality of a parametric loudspeaker. Sinusoidal wave method has been used as the harmonic distortion measurement. In it, the harmonic distortion is measured by analyzing the integral multiplication frequency of a reproduced sine wave. Many measurements by using sine waves with each different frequency are required to measure the wideband harmonic distortions. Therefore, measuring the wideband harmonic distortions with sinusoidal wave method requires much more time. Recently, using Log-TSP (logarithmic time stretched pulse) signal was proposed to measure the wideband harmonic distortions in a short time for non-parametric loudspeakers. Thus in this paper, we attempt to measure harmonic distortion of a parametric loudspeaker by using Log-TSP signal. We carried out an objective evaluation experiment in a soundproof room. The result by using Log-TSP signal was compared with that by using sinusoidal wave method. As a result, we confirmed the result with Log-TSP is equivalent to that with sinusoidal wave method.
The determination of dynamic subtraction for spectral subtraction towards musical tone reduction
Authors: Horii Keisuke , Fukumori Takahiro , Morise Masanori , Nishiura Takanobu , Yamashita Yoichi
Published: April 2012

We should reduce the unwanted noise from noisy speech. Spectral Subtraction (SS) which is one of the noise reduction methods has been proposed by S. F. Boll in 1979, and it can effectively reduce the unwanted noise by utilizing only the observed signal. SS however has a problem that dissonant noise called musical tone is generated after noise reduction. SS estimates the noise with non-speech part and subtracts the estimated noise from observed signal. Flooring process is also performed depend on estimated noise power for supporting the excessive subtraction. Since the musical tone is generated by it, the improvement of SS is required to reduce it. In the past, we proposed SS with the weighted subtraction coefficients in each frequency band for controlling flooring process. In this method, the equal-weighted subtraction coefficients were utilized in every frame, however speech and noise power are different in each frame. To overcome this problem, we newly propose an advanced SS with the color-weighted subtraction coefficients in each frame for effectively reducing the musical tone. Both objective and subjective experiments were carried out for verifying the effectiveness of the proposed SS. As a result, the proposed SS could subjectively reduce the musical tone.
Robustness of the ultrasonic face recognition method to age variation: analysis and experimentation
Authors: Wang Jingyao , Cao Benxi , Wang Yang , Xu Yong , Yang Jun
Published: April 2012

Ultrasonic human face recognition systems transmit ultrasonic Continuously Transmitted Frequency Modulated (CTFM) signal, extract facial geometry characteristic information from the received echo signals, and process the information for face recognition. Compared to vision-based approaches, ultrasonic human face recognition systems reduce the influence of illumination and avoid privacy leak. The existing approaches showed the feasibility of ultrasonic face recognition and achieved acceptable recognition accuracy. However, the aging problem, which equals to performance degradation caused by facial geometry change with time going on, is not considered in the previous researches. In this paper, robustness of the ultrasonic face recognition method to age variation is analysed. A database is built, in which each subject's face information was acquired at intervals of months during the last two years. Based on the database, various recognition experiments are conducted using the different pattern recognition algorithms. Aiming at the experimental results, the analysis to age factor that influences face recognition performance is implemented. The feature extraction methods and the pattern recognition algorithms are developed to increase the recognition rate.
Multi-stage identification for abnormal/warning sounds with onomatopoeia models
Authors: Ogawa Junpei , Hayashida Kohei , Morise Masanori , Nishiura Takanobu , Yamashita Yoichi
Published: April 2012

In recent years, the methods utilizing environmental sounds have been increasingly employed for monitoring the safety of the elder who lives in distant place. Environmental sounds should consist of various sounds in daily life, and identified ones enables to detect abnormity. To detect abnormity, it is therefore required that abnormal/warning sounds are accurately identified among environmental sounds. In the past, environmental sound identification method has utilized acoustic models constructed by each sound source for all environmental sounds. However, as it only stores a few training abnormal/warning sounds, it is difficult to accurately design each abnormal/warning acoustic model. To overcome this problem, we propose a multi-stage identification method for abnormal/warning sounds. In the first stage, it utilizes an abnormal/alarm acoustic sound model and each normal acoustic sound model for detecting the abnormal/alarm sounds with a few training abnormal/warning sounds. In the second stage, it utilizes some acoustic models specializing in onomatopoeia to accurately identify abnormal and alarm sounds. We conducted an evaluation experiment to confirm the effectiveness of the proposed method on identification accuracy of abnormal/warning sounds. As a result, we confirmed that the proposed method was superior to the conventional method in identification accuracy.
A general audience rating surveying system and channel retrieval algorithm based on TV set audio features
Authors: Huang Jingru , Ma Xin , Tian Lan , Du Shibin
Published: April 2012

A general surveying system for audience rating based on audio information are introduced. In this system, the audio signal is sampled from the audio outlet of TV set and high compressed into a special digital package which includes watching timing, audio signal features, and the marks of the TV channel switching. The packaged audio features are high robustness for different types of television sets and different audio volume and have no disturbance of surroundings. In the channel retrieval algorithm, multi-thresholds are used for calculating the correlation coefficients of audio features and matching the audio pattern, and the second searching is adopted for combining match when the retrieved TV channel of the head package is different from the one of the end package. The simulation test results show that the audio recognition rate is above 93%, and for other audios without the TV standard channels can be detected steadily. Keywords: audience rating; audio retrieval; robustness; recognition rate
Acoustic forward scattering by a moving object and its range estimation in littoral experiment
Authors: Lei Bo , Yang Kunde , Ma Yuanliang
Published: April 2012

When an object crosses the source-receiver line, the sound field aberration can be caused by the forward scattering. However, the aberration is difficult to be seen because of the direct blast overwhelming. An experiment was conducted in littoral environment with several hydrophones deployed at different depths. A repeated wideband LFM pulse is transmitted and data is processed with correlation. The experimental results show that the sound field aberration takes minimum values if the object is located mid-point along the source-receiver line, whereas it attains its maximum if the object is close to the source or receiver. The total field is either enhanced or suppressed if the object crosses different Fresnel zones. In addition, the duration of shadow-induced aberration is dependent on the width of the first Fresnel zone, which is longest at the mid-point of the source-receiver line. Furthermore, a range estimation scheme is proposed. The scheme uses two-point field aberrations of the stable arrival caused by forward scattering of moving object. The ranges of intruder are estimated with a prior knowledge of the moving speed, which agree well with the measurements.
Fluctuations of arrival time and amplitude for short-range experimental data
Authors: Duan Rui , Yang Kunde , Ma Yuanliang
Published: April 2012

The arrival time and amplitude fluctuations of about 230-m propagation were analyzed for various source-receiver configurations using experimental data. The minutes-scale and seconds-scale fluctuations of the arrival time were observed on both the direct, surface-reflection and bottom-reflection arrivals while the minutes-scale fluctuation of the arrival amplitude was only observed on the direct arrivals. Cross-correlation coefficients and frequency spectrum of the fluctuation were calculated to explain the causes of the fluctuations. It shows that the movement of the source is an important cause for the fluctuations of the arrival time and the seconds-scale fluctuations of the arrival amplitude. The variability of the ocean structures contributes to both the arrival time and the amplitude fluctuations of the direct arrivals while the sea-surface scattering is the dominant cause for the surface-reflection arrivals. The fluctuation amplitudes of the bottom-reflection and the surface-reflection arrival amplitudes are around 2dB and 7dB, respectively. The fluctuation amplitudes of the direct arrival amplitude range from 1dB to 10dB for different source-receiver configurations.
Temporal coherence of acoustic signals in range dependent background
Authors: Zhang Yin Quan , Wang Ning
Published: April 2012

Temporal coherence of acoustic signals is important for many practical applications, which has been studied both theoretically and experimentally. In previous theories, the fluctuation of sound speed field is assumed to be caused by linear random internal waves and the background is assumed to be range independent. This assumption has two obvious shortcomings: (1) actual fluctuations in shallow water are usually modulated by internal tides; (2) seabed has significant effects on acoustic signal propagation in shallow water, the properties of which is, in general, range dependent (such as topography). In the present presentation, internal tides and seabed properties are involved as the range dependent background, in which the background acoustic field is expressed as coupled mode matrix. A semi-analytic formalism for the temporal coherence of acoustic signal is presented. The dependence of temporal coherence on range and frequency, and the impact due to range-dependence is analyzed. The result is compared with the case range-independent.
The study about through signature of underwater target based on focused beam-forming
Authors: Sheng Xue-li , Liu Yan-qiong , Zhai Chun-ping , Zhu Shi-cai , Liu Yu-dong , Tian Zuo-Xi , Hui Jun-ying
Published: April 2012

In shallow water, the peak and energy of through signature change because of the stack of reflected sound from the sea bottom and sea surface, it will cause much error in signal process. In this paper, the focused beaming-forming method is introduced in through signature processing. Factors of influencing through signature based on focused beaming-forming method are analyzed and the parameters describing through signature are presented. Processing results of actual measurement signal show that if the parameters describing through signature are not affected significantly, it can restrain noise effectively to smooth the through signature curve based on focused beaming-forming method. It also enhances the gain of through signature. Key words: multi-path effects; underwater acoustic image; beaming-forming; through signature; the gain;
Group speed versus phase speed analysis of sound speed fluctuations in a shallow water ocean
Authors: Kuperman W. A. , Cornuelle Bruce D. , Hodgkiss W. S. , Roux Philippe
Published: April 2012

Data collected during eight consecutive hours between two source-receiver arrays in a shallow water environment (Roux et al, J. Acoust. Soc. Am. 124, 3430-3439, 2008) are analyzed using the physics associated with the waveguide invariant. In particular, the use of vertical arrays on both the source and receiver sides provide launch and receive angles in addition to the travel times associated with each eigenray path in the waveguide. From travel times and source-receiver angles, each eigenray amplitude is projected into group velocity-phase velocity (vg-vp) space for each acquisition. The time evolution of the vg-vp representation during the 8-hour long experiment is discussed. Group speed fluctuations observed for a set of eigenrays with upper turning points at or near the thermocline are compared to independent sound speed measurements obtained by a CTD chain at two locations in the area.
Variability of the waveguide invariant in a range independent shallow-water environment
Authors: Cockrell Kevin
Published: April 2012

While it is generally true that the waveguide invariant β is approximately equal to one in many shallow-water environments, the value of β observed in acoustic intensity striation patterns often deviates from the assumed value of one by 30% or more. The precise value of β, like the acoustic field itself, depends on source frequency, source and receiver depths, sound speed profile, range, etc. For example, previous work has found that the observed value of β in shallow water can deviate from the typical value of one if the source and receiver are located below the thermocline, because in that case the acoustic intensity is strongly affected by lower order modes which do not have β≈1 [D. Rouseff, Waves in Random Complex Media, 2001]. This talk will further explore the dependence of β on the source and receiver depths, and on the sound speed profile by using the WKB approximation. Because the group of modes which dominates the acoustic intensity depends on range and frequency, the dependence of β on range and frequency will also be explored.
Waveguide invariant and dedispersion transform
Authors: Dazhi Gao
Published: April 2012

The waveguide invariant is described as a single scalar parameter for a given waveguide environment. The notion of waveguide invariant has been applied widely in under water acoustic source ranging, beamforming etc. A Fourier-like transform called to dedispersion transform, which can remove the dispersion of multi-modes at the same time, was proposed in our previous paper (“Ning Wang, 9th. Western Pacific Acoustic Conference Beijing 2009”). In this presentation, we refine the relationship between the waveguide invariant and the dedispersion transform, and the dedispersion transform is extended to low frequency region and typical summer waveguides. Numerical simulations and several real-data processing will be also reported.
The contribution of interaural time and level differences to the precedence effect at high frequencies
Authors: Seeber Bernhard U.
Published: April 2012

The precedence effect (PE) allows us to locate sound sources correctly in rooms despite the presence of interfering reflections. It has been shown to function at high frequencies with highly modulated stimuli. These studies were done in the free-field where interaural time (ITDs) and level (ILDs) differences are in their natural combination. The present study investigated the relative contribution of ILDs and ITDs to the PE with high-frequency zero-phase harmonic complex tones. A localization dominance task was used in which participants indicated the location of the lead-lag stimuli and judged if sounds were perceived as fused. A preliminary analysis indicates that the PE emerged when either ITDs or ILDs were applied to lead and lag stimuli while the other binaural cue was held at zero. Patterns for localization dominance and fusion were nearly identical for ITD and ILD conditions, suggesting that ITDs and ILDs were equally effective for these highly modulated stimuli. Fusion of lead and lag extended to somewhat longer delays with smaller cue magnitudes, i.e. the more binaural cues differed between lead and lag the more likely they were to be segregated. The results support the idea that PE mechanisms are similar for ITDs and ILDs.
Echo thresholds for reflections from acoustically diffusive architectural surfaces
Authors: Robinson Philip W. , Walther Andreas , Faller Christof , Braasch Jonas
Published: April 2012

Diffusive architectural surfaces play an important role in performance venue design for architectural expression and proper sound distribution. However, previous psychoacoustic research on perception of reflections and the precedence effect has focused on specular reflections. This study compares the echo threshold of specular reflections, measured using an adaptive up-down method with music and speech stimuli, against those for reflections from realistic architectural surfaces, and against synthesized reflections that isolate individual qualities of reflections from diffusive surfaces, namely temporal dispersion and spectral coloration. It is found that temporal dispersion up to 16ms in the reflection response, and peak amplitude reduced as much as 18.5 dB results in an echo threshold shorter than that for a specular reflection of comparable amplitude. Rather, the threshold is comparable to that of a specular reflection of similar energy. This indicates that the auditory system integrates the temporally dispersed energy into a single stream.
A binaural model that uses head-movements to evaluate acoustical spaces
Authors: Braasch Jonas , Clapp Samuel , Parks Anthony , Xiang Ning
Published: April 2012

Binaural models have a long tradition in the instrumental analysis of acoustical spaces. Room acoustical parameters such as the Binaural Quality Index (BQI) are derived directly from the measured Binaural Room Impulse Response (BRIR) of a concert space. The BRIRs are measured using an artifical head with a fixed head position and consequently cannot show the effect of head movements, which are essential for human listening performance. Based on a novel model architecture that utilizes head movements [Braasch et al., 2011, J. Acoust. Soc. Am. 129, 2486 (A)] and psychoacoustic experiments, the effect of head movements on the perceptual judgment of room acoustical parameters will be discussed. In addition, BRIRs for different azimuth angles are calculated from higher-order ambisonic microphone data that were obtained in several concert halls in the Northeastern United States. [Clapp et al., 2011, J. Acoust. Soc. Am. 130, 2418 (A)]. It will be demonstrated how the standard acoustical parameters change with head orientation and how dynamic head-movement cues can be utilized to better predict the perceived quality of concert spaces.
The desire for decorrelation—applications from the recording studio
Authors: Case Alexander
Published: April 2012

Multitrack production solves challenges of masking, localization, and intelligibility while pursuing aesthetics associated with reverberance, distance, envelopment, and source width through any means available in the signal processing-rich environment of the recording studio. Contemporary sound recording techniques are presented that might influence the design for achieving similar results through architectural signal processing.
Ultrasonic hearing in frogs: inner ear morphological correlates
Authors: Narins Peter
Published: April 2012

Three species of anuran amphibians (Odorrana tormota, O. livida and Huia cavitympanum) have recently been found to detect ultrasounds. We compared morphological data collected from the ultrasound detecting species with data from Rana pipiens, a frog with a typical anuran upper cut-off frequency of ca. 3 kHz. In addition, we examined the ears of two species of Lao torrent frogs, O. chloronota and Amolops daorum that live in acoustic environments resembling those of the ultrasonically sensitive frogs. Our results suggest that the three ultrasound-detecting species have converged on small-scale functional modifications of the basilar papilla (BP), the high-frequency hearing organ in the frog inner ear. These modifications are also seen in the ears of O. chloronota, suggesting that this species is a candidate for high-frequency hearing sensitivity. These data form the foundation for future functional work probing the physiological bases of ultrasound detection by a non-mammalian ear. Supported by NIDCD DC-00222, Paul S. Veneklasen Research Foundation, and the UCLA Academic Senate (3501).