bluuebunny/crossref_metadata_2025 huggingface dataset:
Time history theoretical analysis for stick-slip vibration of violin strings
Authors: Utsuno Hideo
Published: November 2016

The stick-slip vibration of violin string was completely formulated by using CA (Cellular Automaton) method in time history analysis. The local neighborhood rule of the CA method was derived for the travelling wave along the string, both ends of string, and the bow point. The key in success is that the bow point is to be fixed point and also to be velocity excited point at the same time. This means that reflected wave velocity is equal to the incidental wave velocity plus bow velocity. Displacement of arbitrary point of string is calculated and is compared with measured one. Excellent agreement of displacement suggests that CA method can simulate the stick-slip vibration of violin string. Acceleration time of the bow until it reaches a certain speed is also studied to form a complete Helmholtz waves. Step input and acceleration time within fundamental period is not valid to form Helmholtz wave. Acceleration time of more than four times of the fundamental period can substantially form Helmholtz waves.
Numerical investigation on the influence of wall vibrations on the behavior of the lip excitation in brass instruments
Authors: Fréour Vincent , Masuda Hideyuki
Published: November 2016

The influence of wall vibrations in brass instruments has been an object of substantial debate in both musician and scientific communities. The excitation of the instrument wall (from the acoustic field inside the instrument and through the direct mechanical excitation from the lips) may influence the acoustic pressure field inside and outside the instrument, potentially resulting into some noticeable effects on the acoustic input impedance and radiated sound. These effects have been the object of numerical and experimental studies on different families of wind instruments. Furthermore, these vibrations are also likely to induce some perturbations on the excitation mechanism itself (the lips of the player), that may contribute to variations of the produced tone, as well as influence the sensations of the player. In this paper, the effects of the this mechanical coupling between the lips and vibrating wall is studied numerically using a physical model of the lips, coupled acoustically to the air-column of the instrument, as well as mechanically with the instrument wall described by a simplified mechanical admittance. Results from time-domain simulations and linear stability analysis will be presented. Some pathological situations, where a structural mode is close to the playing frequency will be particularly discussed.
Precision harmonics simulation for vacuum-tube electric-guitar pedals and preamplifiers
Authors: Takemoto Kanako , Oshimo Shiori , Hamasaki Toshihiko
Published: November 2016

Recently, vacuum-tube pedals have attracted many electric guitar players, though solid-state circuits or even digital processors are widely accepted. The purpose of this study is to clarify the origin of distorted timbre of the vacuum-tube pedal by circuit simulation with a novel vacuum-tube model. In general, the specific sound of vacuum tube for both hi-fi and electric guitar amplifier is attributed to the even number of harmonics originated from the inherent non-linearity. However, significant difference exists in the input signal amplitude between hi-fi and electric guitar system. The simulation with a conventional non-linear model has been limited within the regulated signal swing of hi-fi audio system. The dynamic non-linear phenomena for large signal, such as an input signal modulation, are not considered. Thus, unrealistic harmonics appear in the pedal simulation due to discontinuity of model parameters. The new model is obtained by physical analysis based on enough measurements, considering the actual input condition of pedal and preamplifier as well. The key property is a seamless modulation of non-linearity influenced by circuit operation. As a result, we have succeeded in simulating changes of the dynamic harmonics accurately for the first time.
Changes to soundscapes at outdoor gathering spaces in Fukushima, Japan, caused by the severe nuclear power plant accident
Authors: Nagahata Koji
Published: November 2016

Five years have passed since the accident at the Fukushima Daiichi Nuclear Power Plant, which caused changes to daily life in the city of Fukushima, Japan. These daily life alterations caused changes to the soundscapes of Fukushima, which are still evolving now. In this study, the soundscape change at Fukushima’s outdoor gathering spaces is discussed using field recordings by the author. Shortly after the accident, few human voices could be heard at outdoor gathering spaces, but natural sounds such as birdsongs could be heard as usual per the time of year. Park decontamination started during summer 2011. In some parks where decontamination was successfully completed, people’s voices and sounds of children playing returned during spring 2012. However, in other parks where decontamination was done but ineffective, the lack of human voices and artificial sounds continued until radiation levels decreased sufficiently. In this way, soundscape change at outdoor gathering spaces represents people’s attitude toward the radioactive contamination; soundscape recording documents not only sonic environments but also people’s lives.
E-participation in the context of the evaluation of the sonic environment
Authors: Fiebig André , Schulte-Fortkamp Brigitte , Genuit Klaus
Published: November 2016

E-participation methods are increasingly applied to promote public participation and to achieve greater awareness and involvement by citizen by providing the opportunity to directly report to, e.g., governmental organizations. Within the framework of the German Year of Science 2015, a special initiative, called “Sound of the City,” was carried out to encourage the public to report on how their cities typically sound. Participants could upload audio recordings of acoustical environments and had to relate the respective audio files to the corresponding places. Moreover, the participants could additionally describe the uploaded sounds. This action lasting several months led to a comprehensive database of audio files including related places, descriptions, and ratings. The database was systematically analyzed with respect to several issues, such as general participation level, hot spot identification, and common features of uploaded sounds. The paper will highlight the benefit and limitations of such actions and initiatives applying e-participation with respect to community noise issues and urban planning. [The “Sound of the City” initiative took place within the framework of the Year of Science 2015 and was organized and supported by the Federal Ministry of Education and Research (BMBF).]
On-site and laboratory soundscape evaluations of three recreational urban spaces
Authors: Sørensen Anna Josefine , Larsen Thea Mathilde , Bjerre Lærke Cecilie , Santurette Sébastien , Jeong Cheol-Ho
Published: November 2016

Soundscape quality was evaluated using four subjective psychological rating factors in three recreational urban spaces in which water and a variation of other natural and anthropogenic sound sources were present. The noise level was measured at each site during occupant peak flows and recordings for listening experiments were made simultaneously. Listeners answered questionnaires either on site or following playback of the recordings in the laboratory, with or without access to each site’s visual context. They rated their perception of loudness, acceptance, stressfulness, and comfort, along with their preference toward eight sound sources. The comfort ratings were negatively correlated with loudness and stressfulness and positively correlated with acceptance. The sound level was found to be a good predictor of these subjective parameters in the laboratory, but not on site. Moreover, the availability of the visual context in the listening experiment had no effect on the ratings. The presence of trees and water was also found to increase on-site comfort. Generally, the participants were more positive towards natural sound sources on-site. Overall, the results suggest that on-site context plays an important role for evaluating acoustic comfort in urban recreational areas.
Evaluation of the noise disturbance and noise limits of a riding horse shooting competition
Authors: Montoya Juan C.
Published: November 2016

Shooting ranges have turned into an unwanted activity in rural communities due to high levels of impulse sound pressure levels. Permanent complaints from neighbors or neighborhood associations pertaining health concerns and noise pollution around areas nearby shooting ranges are a driving force to create state noise control laws. As such, examining zoning will limit (restricting licenses) to the use of agricultural or residential land to be used for shooting events. Gun noise is measurable and therefore is able to be quantified and characterized. This study evaluates with field noise measurements the events from a “shooting ranch” which currently operates offering a shooting competition while riding a horse emulating the Old West. The analysis of field data pretends to look at metrics such as LAeq (A-weighted equivalent sound level), L90 (noise level exceeded for 90% of the measurement period), and Lmax (maximum level if the measurement period from 50 caliber single action revolver gun sounds while shooter is in movement riding a horse. The field measurements were conducted in different nearby residences. The impulse noise lead to examine instead individual gunshots which exceeded background levels by 14 to 34 decibels. Noise levels from Shooting Ranch events are sometimes more than three times the background noise level. Leq and L90 metrics are applicable within individual events.
Influence of launch platform cut-outs on flow and acoustic behavior of rocket exhaust
Authors: Natarajan Karthikeyan , Venkatakrishnan Lakshmi
Published: November 2016

A launch vehicle experiences intense acoustic loading in the initial phase of its lift-off which affects the launch vehicle structure, sensitive payloads, and electronics on board. There is immense interest in alleviation of acoustic loads resulting in reduced need for strengthening of the vehicle structure. The effect of jet blast deflector shape on the acoustic loading has been extensively investigated, both computationally and experimentally, by simulating jet(s) impinging on a flat plate. However, contributions from the launch vehicle environment, such as the launch platform, are often ignored. The motivation for this study is that the flow over the launch platform is likely to be significantly influenced by the cut-outs made in the platform for the nozzles. As the nozzles emerge out from the cut-outs during lift-off, the jet exhaust grows and interacts with the launch platform, contributing to the overall acoustic loads experienced by the vehicle. This paper presents an experimental investigation of rocket exhaust interaction with the launch platform using single and twin jets impinging on a flat plate with cut-outs. The measurements include high speed shadowgraphy and microphone measurements in the near and far-field to enable flow and acoustic characterization.
A design method of fairing acoustic noise mitigation for Korean space launcher
Authors: Park Soon-Hong , Seo Sang-Hyun
Published: November 2016

In this presentation, fairing acoustic protection system (APS) for Korean space launcher is introduced. The design of APS for payload fairing of KSLV-II is summarized. The current APS consists of acoustic blankets for midfrequency range absorption and acoustic resonators for low frequency noise absorption. Detailed design procedures of APS, which include external acoustic loading prediction based on the modified NASA SP-8072 monograph and fairing system vibro-acoustic analysis, are presented. An acoustic test of a cylindrical composite structure with APS is also presented. The effect of spatial correlation of external acoustic excitation on the vibro-acoustic response of fairing structure and its interior volume is discussed. The concept of future APS by using the micro-perforated panel absorber combined with Helmholtz resonators is also introduced. Effects of high pressure environment on the acoustic absorption of micro-perforated panel is discussed.
Pupillary responses signify more than just effort: Windows into processing, prediction, reflection, and uncertainty
Authors: Winn Matthew
Published: November 2016

Pupil dilation has long been used to index effort or general cognitive arousal in a variety of tasks. Traditionally, the main outcome measure is peak pupil dilation or the mean pupil dilation within some target window. In this talk, we explore how the time course of the growth and decay of pupil dilation can lend further insight into auditory processing. A series of pupillometry experiments revealed elevated listening effort resulting from various influences such as poor spectral resolution, unpredictability of sentence content, selective attention, or the need to reflect on what was heard to produce a creative response. Peak and mean pupillary responses generally confirmed original hypotheses, but more interesting findings emerge upon inspection of unforeseen trends in the data. Specifically, the timing of the pupil decay response can arguably be interpreted as a signature of the completion of sentence prediction, or the completion of processing that relies on memory trace (rather than perception) of the signal. The ultimate goal of this discussion is to unpack a new approach to pupillometry experiments that is aimed at separating processing that stems from perception versus processing that stems from cognitive processing, or reflecting upon a memory trace of the stimulus.
An objective measure of temporal fine structure correlates with perception in a speech discrmination task
Authors: McAlpine David , Bowen Macarena , Undurraga Jaime
Published: November 2016

Sensitivity to the temporal fine structure (TFS) of sounds is considered important for understanding speech, especially in the presence of background noise, but few studies have employed objective measures to assess the relationship between sensitivity to TFS and listening to speech. Here, we used EEG (electroencephalography) to obtain an objective measure of sensitivity to the TFS of low-frequency sounds (520-Hz amplitude-modulated tones) in listeners who also undertook a speech-discrimination task (discrimination of vowel-consonant-vowel, VCV, sounds in background noise). Two different sound levels (60 and 80 dB SPL) and three different signal-to-noise ratios (SNRs; -6, 0 and + 6 dB) were used. Three symmetric interaural phase modulations (IPMs), were measured: -60°/60°, -90°/90°, and -120°/120°. A fourth condition used an IPM switching between 0° and 180° degrees. IPM-FRs were higher for the 0/180° degrees than all other conditions. Employing the ratio of the 3 symmetric to the asymmetric (0/180°) IPM-FR as a measure of the neural dynamic range of TFS processing, we found that VCV scores increased with age at 60 dB SPL, but not 80 dB SPL, and indeed were better at 60 compared to 80 dB SPL in older listeners, and correlated with a higher ratio of IPM-FRs.
Breaking time reversal symmetry with coriolis mean flow systems
Authors: Farzbod Farhad , Leamy Michael J.
Published: November 2016

One way to break time reversal symmetry is to have a moving wave propagation medium. If the acoustic wave vector and the moving fluid velocity are collinear, such as the one created by Fluery et al. (Science, 343, 2014) we can use the wave vector shift caused by the fluid flow to break reciprocity. An alternative approach we have taken is to use a fluid velocity field which enters the differential equation of the system as a cross product term with the wave vector. A circular field in which the fluid velocity increases radially has a Coriolis acceleration term. In this system, the acoustic wave enters from the central wall and exits from the perimeter wall. Equations of conservation of mass and momentum, after linearization can be simplified to: (∂/∂t+V.▽)P+ρC2▽.(u+V) = 0, ρ∂/∂t(u+V)+ρ(V.▽)u+ρ(u.▽)V+ρ(V.▽)V+▽P = 0, in which u is the particle acoustic velocity and V = rωeθis the fluid velocity due to the rotational velocity field. It can be shown that the third term in the second equation can be simplified as:ρωek×u. This term Coriolis acceleration induces nonreciprocity. In this work, we solved the differential equation numerically and investigated the effect of fluid velocity on the nonreciprocity factor.
Quantised acoustic meta-surfaces using metamaterial bricks
Authors: Memoli Gianluca , Caleap Mihai , Asakawa Michihiro , Sahoo Deepak , Drinkwater Bruce W. , Subramaniam Sriram
Published: November 2016

Acoustic meta-surfaces offer an extremely successful way of transforming an input acoustic wave into almost any diffraction-limited field. This is typically achieved by encoding a phase delay function on the unit cells that form the meta-surface, which is then manufactured by rapid prototyping. One of the limits of current technology is that meta-surfaces built in this way are targeted to a single, specific application. In this work, we present a method of building acoustic meta-surfaces using metamaterial bricks, designed to operate at 40 kHz and quantized both in the space and in the phase domains. First, we present finite-elements simulations and experiments to discuss key parameters of our bricks as frequency response, directivity, impedance. Second, our metamaterial bricks are used here to achieve some key beam transformations like steering and focusing, but also more complex fields like an acoustic tractor beam. We use the latter to levitate one or more polystyrene beads and diffraction theory to explain some of the findings. Our method, where the phase in a specific position may be changed by changing the specific brick, may find applications in loudspeaker design, ultrasound imaging/therapy, or acoustic particle manipulation.
Articulatory settings of high- and low-proficiency second-language speakers
Authors: Wilson Ian
Published: November 2016

Articulatory setting, a language-specific underlying position of the articulators during speech, can be measured by using inter-speech posture (ISP)—the speech rest position. In an x-ray study, it was shown to behave like a target posture, in that it has as little variability as /i/, one of the least variable vowels. In an rt-MRI study, it was found to be more mechanically advantageous than absolute rest position. In an ultrasound/Optotrak study, results showed differences across normalized monolingual groups of speakers, and bilinguals who were perceived to be native in each of their languages had a different articulatory setting for each language. In an EMA study, tongue width in the coronal plane during ISP was found to be less efficient in second-language (L2) speakers of English. In recent research using ultrasound and lower-proficiency Japanese-English bilinguals, we were surprised to find differences in their articulatory settings for each of their languages, when it would have been simpler for such a speaker to use the articulatory setting of their dominant language while speaking their non-dominant one. In this talk, a state-of-the-art analysis of research on articulatory setting will be given including recent research on Japanese-English bilinguals.
Hailu Hakka and Taiwan Min tone changes
Authors: Pan Ho-Hsien , Lyu Shao-Ren , Huang Hsiao-Tung
Published: November 2016

A pitch contour overlap and a tone-sandhi relationship can both contribute to confusion in tone that lead to change in tone. This study investigates the duration, f0, and phonation between Hailu Hakka and Taiwan Min tone pairs that (1) were neither overlapped in pitch contours nor tone-sandhi related, such as Hakka tones 55 and 33, and Min tones 55 and 13; or (2) were overlapped in pitch but not tone-sandhi related, such as Hakka 33 and 11, and Min 13 and 31; or (3) were not overlapped in pitch but tone-sandhi related, such as Hakka and Min tones 3 and 5, or (4) were both overlapped in pitch contour and tone-sandhi related, such as Hakka 24 and 33, and Min tones 31 and 33. The results indicated that Hakka speakers under 30 years of age produced tones 24 and 33 that were overlapped in pitch and tone-sandhi related, and with similar pitch contour; whereas, Min speakers, regardless of age, produced tones 3 an 5 that were not overlapped in pitch but were sandhi-related bi-directionally (3↔5) similarly. It is proposed that sound changes occur among sandhi related tones with and without overlap in pitch contours.
A brief introduction to distributed compressed sensing and its typical reconstruction limits
Authors: Shiraki Yoshifumi , Kabashima Yoshiyuki
Published: November 2016

A brief introduction to compressed sensing and a discussion of its reconstruction limit analysis are presented. In addition, we briefly introduce the replica method, a technique in statistical mechanics, to show its effectiveness for typical performance analysis on signal processing problems. In this presentation, we especially focus on a signal model called joint sparse model 2 (JSM-2) or the multiple measurement vector problem, in which all sparse signals share their support. The signal model JSM-2 is important for dealing with practical processing problems. We investigate the typical performance of l2,1-norm regularized minimization and the Bayesian optimal reconstruction schemes in terms of mean square error for noiseless and noisy measurement JSM- 2 problems. Employing the replica method, we show that these schemes, which exploit the knowledge that the signal support is shared, can recover the signals more precisely as the number of channels increases. The main focus of our presentation is theoretical analysis; however, we will introduce applications of the signal model JSM-2 for acoustic problems. Most of our presentation is based on papers J. Stat. Mech. (2015) P05029 and J. Stat. Mech. (2016) 063304.
Complex non-negative matrix factorization: Phase-aware sparse representation of audio spectrograms
Authors: Kameoka Hirokazu , Kagami Hideaki
Published: November 2016

Non-negative matrix factorization (NMF) has attracted a lot of attention after being proposed as a powerful approach for audio source separation. Factorizing the magnitude spectrogram of a mixture signal into the product of two non-negative matrices can be interpreted as approximating the observed spectra at each time frame as a linear sum of basis spectra scaled by time-varying amplitudes. One drawback of this approach is that the additivity of magnitude spectra is assumed, which holds only approximately. To address this drawback, we introduce a novel framework called the “Complex NMF,” which makes it possible to realize NMF-like signal decompositions in the complex spectrogram domain. We further develop a time-domain extension of this approach, which is noteworthy in that it allows for several extensions that were not possible with the conventional NMF framework.
Decomposing guided wavefields with dictionary learning
Authors: Harley Joel B. , Alguri K. Supreet
Published: November 2016

Ultrasonic guided waves have become a powerful tool for the structural health monitoring of large structures due to their low attenuation and high damage sensitivity. Yet, one significant challenge of working with guided waves is their complexity. Ultrasonic guided waves demonstrate strong multi-modal and dispersive characteristics and are often used in structures with many boundaries and reflecting elements. In prior work, compressive sensing theory and sparse recovery algorithms have been used to learn the multi-modal and dispersive properties of guided waves based on relatively simple wave models. The algorithms can predict and construct wavefields based on these learned properties. Furthermore, this predictive power can be used to detect and locate damage in a structure. However, these algorithms cannot yet predict and utilize reflections as well as other complexities (such as inhomogeneities) in a structure due to the simplicity of the current model. This paper presents a dictionary learning approach to learn this model directly from simulation or experimental data. This paper focuses on studying the models (i.e., dictionaries) created by dictionary learning under various simulated conditions. We demonstrate that we can use these dictionaries to accurately predict wavefields when waves are distorted by reflections, velocity inhomogeneities, and velocity anisotropy.
Acoustic single-pixel imaging using compressive sensing
Authors: Rogers Jeffrey S. , Naify Christina J. , Rhode Charles , Edelmann Geoffrey F.
Published: November 2016

The acoustic analog to the single-pixel imager found in the optical and THz imaging literature is presented. Here, the single-pixel detector is replaced with a single omnidirectional acoustic receiver and combined with a series of masks that allow for multiple orthogonal measurements. Acoustic source localization is performed based on the theory of compressive sensing which allows for the resolution of N acoustic sources using M measurements where M << N. A theoretical model for the acoustic single pixel imager is presented and compared to finite element models (FEM) simulated in COMSOL and experimental data taken in an air acoustic 2D waveguide. It is shown that the acoustic single pixel imager is capable of localizing multiple targets using only a single omnidirectional acoustic receiver. [Work supported by the Office of Naval Research.]
Grid-free channel estimation of acoustic measurement data from SAVEX15
Authors: Choo Youngmin , Park Jungyong , Park Yongsung , Park Minseuk , Seong Woojae
Published: November 2016

Acoustic channel impulse responses (CIRs) consisting of multiple arrivals are extracted from the measured data during SAVEX15 experiment (conducted in the northern East China Sea during May 2015) using compressive sensing, providing high-resolution results due to sparsity promoting objective function. Performance of grid-based compressive channel estimation degrades when basis mismatch occurs because delays of the arrivals do not correspond to delays on the grid. To overcome this, grid-free compressive channel estimation is derived from the formulation of grid-free compressive beamforming [J. Acoust. Soc. Am. 137, 1923-1935 (2015)]. The grid-free compressive channel estimation is applied to signals from SAVEX15; the signals were transmitted from a towed source and received at two separate vertical line arrays (16 elements each); they are used to measure the signals traveling through waveguides. Acoustic structures are observed clearly due to high-resolution CIRs along depth, which are varied by the moving source and internal wave activities.
Personal audio reproduction using two wearable end-fire loudspeaker arrays
Authors: Imaizumi Kenta , Haneda Yoichi
Published: November 2016

We propose a wearable personal audio reproduction system that uses two end-fire loudspeaker arrays. The two loudspeaker arrays are attached in front of the listener’s breast. Consequently, the system is not close to the ears. Further, because the main beams of the two arrays are directed at each ear, crosstalk between the left and right signals is low. In addition, sound leakage is also low as a result of directivity control. We constructed a prototype of the system using four loudspeaker elements for each array, with intervals of 4 cm and array length 12 cm. The attenuation level of the area at approximately 50 cm away from the end-fire arrays was found to be -20 dB sound pressure level (SPL) relative to the main beam pointed at the ear. In addition, the crosstalk SPL difference was approximately 12 dB. We can control the sound image position by convolving the head-related transfer function with the sound source. Further, no crosstalk canceller is needed because the proposed method secures the interaural intensity difference. Improvement or application of the proposed system can facilitate realization of a personal sound field.
Relationship between speech recognition performance and related spoken document retrieval performance
Authors: Kanedera Noboru
Published: November 2016

Along with multimedia information including the speech information having increased, spoken document retrieval has attracted attention. Keywords or query sentences are used for spoken document retrieval. The keywords or query sentences cannot be used when we want to watch the video scenes which are related to a watching video scene. In this report, we propose a method of evaluating spoken document retrieval between video scenes, and report the results of the investigation about the relationship between the speech recognition performance and spoken document retrieval performance by using watching video scene. As a result of simulation experiment, it was confirmed to affect the search performance in order of substituted error, deletion error, insertion error, in the search both by question sentence and by the scene. The search by the scene is easy to be affected by the speech recognition error than the search by the question sentence. It was found that the influence of the insertion error in the search by the scene was bigger than the search by the question sentence.
An evaluation of speech reconstruction for degraded speech of optical microphone with deep neural network
Authors: Mizuno Tomoyuki , Fukumori Takahiro , Nakayama Masato , Nishiura Takanobu
Published: November 2016

Measuring distant-talking speech with high accuracy is important in detecting criminal activity. Various microphones such as the parabolic and shotgun microphones have been developed for measuring it. However, most of them have difficulty in extracting distant-talking speech at a target position if they are surrounded by noisy sound sources. Therefore, this study focuses on a microphone system to extract the distant-talking speech by vibrating a papery object using laser light. This system is referred to as an optical microphone in this study. The sound quality of the optical microphone is especially degraded at higher frequencies because it utilizes an external diaphragm consisting of various materials as the vibrating papery object. In this study, we therefore propose a reconstruction method for degraded distant-talking speech observed with the optical microphone. The method is realized by using a deep neural network (DNN) that is trained as the system between clean and observed speech signals. In the proposed method, the log-power spectra of 11 frames are used for the input of the DNN. Finally, we confirmed the effectiveness of the proposed system through an evaluation experiment.
Fast multiple moving sound sources localization utilizing sparseness of speech signals
Authors: Sato Eiji , Tatekura Yosuke
Published: November 2016

We propose multiple moving sound sources localization with reduced operation time utilizing the sparseness of speech signals, and we demonstrate the efficacy of proposed method in actual environment experiments. Sound source localization is an essential function for tasks such as robot audition, and particularly for sound sources that often move in the actual environments. Sound source localization requires high resolution and real-time processing because they are utilized for post-processes such as sound source separation. The proposed method is based on MUSIC (MUltiple SIgnal Classification) which is known as a sound source localization method with high resolution. However, MUSIC has a large computational complexity. When the signal has large energy over a wideband in the frequency domain, the operation time increases because MUSIC estimates the direction of the sound source at every frequency. Therefore, we attempted to reduce the operation time by using the sparseness of speech signals in which the sound energies exist sparsely in the time-frequency domain. We chose frequencies with large power from frequency characteristics of the observed signals. Sound source localization was performed in the chosen frequencies. The operation time of the proposed method was about 8.1 times faster than the case of utilizing all frequencies.
Correct phoneme sequence estimation using recurrent neural network for spoken term detection
Authors: Sawada Naoki , Nishizaki Hiromitsu
Published: November 2016

This paper proposes a correct phoneme sequence estimation method that uses a recurrent neural network (RNN)-based framework for spoken term detection (STD). It is important to reduce automatic speech recognition (ASR) errors to obtain good STD results. Therefore, we use a long short-term memory (LSTM), which is one of an RNN architecture, for estimating a correct phoneme sequence of an utterance from phoneme-based transcriptions produced by ASR systems in post-processing of ASR. We prepare two types of LSTM-based phoneme estimators: one is trained with a single ASR system's N-best output and the other is trained with multiple ASR systems' 1-best outputs. For an experiment on a correct phoneme estimation task, these LSTM-based estimators could generate better phoneme-based N-best transcriptions rather than the best ASR system's ones. In particular, the estimator trained with multiple ASR systems' outputs worked well on the estimation task. Besides, the STD system with the LSTM estimator drastically improved STD performance compared to our previously proposed STD system with a conditional random field-based phoneme estimator.
Evolutionary optimization of long short-term memory neural network language model
Authors: Tanaka Tomohiro , Moriya Takafumi , Shinozaki Takahiro , Watanabe Shinji , Hori Takaaki , Duh Kevin
Published: November 2016

Recurrent neural network language models (RNN-LMs) are recently proven to produce better performance than conventional N-gram based language models in various speech recognition tasks. Especially, long short-term memory recurrent neural network language models (LSTM-LMs) give superior performance for its ability to better modeling word history information. However, LSTM-LMs have complex network structure and training configurations, which are meta-parameters that need to be well tuned to achieve the state-of-the-art performance. The tuning is usually performed manually by humans, but it is not easy because it requires expert knowledge and intensive effort with many trials. In this study, we apply covariance matrix adaptation evolution strategy (CMA-ES) and automate the tuning process. CMA-ES is one of the most efficient global optimization techniques that has demonstrated superior performance in various benchmark tasks. In the experiments, the meta-parameters subject to the tuning included unit types at each network layer (LSTM, RNN, and feed-forward), the number of hidden layers, mini-batch size, and so on. We show that CMA-ES successfully optimizes the meta-parameters of LSTM-LM improving the recognition performance. We also report what are the common characteristics of the well-performing models by analyzing the results of the evolutions.
Development of a communication system using audio signal using audible frequency bands
Authors: Terayama Chihiro , Niitsuma Masahiro , Yamashita Yoichi
Published: November 2016

This paper proposes a technique of embedding information into audio signal to communicate using loudspeakers and microphones mounted on tablet-type devices. This technique is based on amplitude modulation for high audible frequency bands. The embedded information is represented by existence of sine waves with 11 different frequencies. In consideration of the frequency characteristics of tablet-type devices, two frequency bands to embed information, were chosen: (1) frequencies up to 12 kHz; (2) frequencies up to 16 kHz. The experimental result showed 85.9% of correct decoding in the case of (1), and 76.8% in the case of (2). Moreover, less audible noise was perceived in the case of (2).
The impact of architecture on both the intelligibility and privacy of speech
Authors: Roy Kenneth P.
Published: November 2016

Architects design the “visual space” that we can all see, and this we call architecture. Acousticians upon entering this visual space invariably clap their hands, and as such are assessing the resulting “aural space.” Acoustics is the result of architecture, and so, speech clarity is determined by architecture—meaning the size, shape, and surface treatments chosen by the architect to deliver the mission for which the space is intended. We can look at (model) or listen to (field evaluate) the impulse response of a room, and from this determine the degree of speech clarity that is projected to any listening location. Designing for good speech intelligibility within a space means providing adequate speech clarity and then protecting this signal with a high signal-to-noise ratio (S/N) to ensure that the signal can be heard above the ambient noise. If on the other hand we seek to limit the speech intelligibility (privacy) between spaces, we need to design the architecture to limit the intrusion of speech between spaces taking into consideration the architectural choices as barriers to sound transmission via walls, ceilings, and floors. Examples of design approaches in both cases of intelligibility and privacy will be shown.
Effects of dominance of the first-arriving sound on speech intelligibility in reverberant sound fields
Authors: Sato Hayato , Morimoto Masayuki , Hoshino Yasushi
Published: November 2016

Generally speaking, the energy of the direct sound exceeds that of reflections in rooms for listening to speech. In this case, early-reflections enhance loudness of the direct sound. It is well-known that the useful-to-detrimental sound ratio, in which useful is the sum of the direct sound and early-reflections and detrimental is late-reflections, can predict speech intelligibility. On the other hand, considering speech leakage from the adjacent room, the first-arriving sound is attenuated by the boundary wall, and it is sometimes too weak to trigger the temporal integration of loudness. This would change the time structure of the useful energy. In the present study, prediction of speech intelligibility in such particular situations was investigated. Word intelligibility tests were performed in reverberant sound fields varying reverberation time, direct-to-reverberation ratio, and rise time at the onset of reverberation sound. The results demonstrated that prediction accuracy of the useful-to-detrimental ratio was still in a practical range though the parameters systematically affect the prediction error.
Preliminary analysis of social calls used by tagged humpback whales in the Los Cabos, México, breeding ground
Authors: Seger Kerri , Zoidis Ann M. , Hawkins Brooke L. , Jimenez-Lopez M. Esther , Thode Aaron M. , Urban R. Jorge
Published: November 2016

Humpback whales produce a large variety of diverse sounds beyond their well-known songs. Mothers, calves, and non-breeding whales may use these “social sounds” to maintain group cohesion, facilitate feeding, and/or increase a calf’s safety. To date, social sounds have been studied off Hawaii, Alaska, and Australia. During the 2014-2015 breeding seasons, acoustic tag data were collected from humpback whales off Los Cabos, México. Twenty-one tags successfully recorded data (nine in 2014; 13 in 2015) from three mother/calf pairs, ten mother/calf/escort groups, and eight competitive pods varying in size from four to fourteen individuals. A subsequent manual analysis found 1587 social sounds in 2258 total minutes of data. Currently, there are 42 identified distinct social sounds used by humpback whales in the Los Cabos breeding ground, six of which seem unique to the Cabo region as compared to those published from Australia, Hawaii, and Alaska. Call type usage, call rates, and repertoire diversity (measured using information entropy) vary between mother/calf, mother/calf/escort, and competitive groups. Results suggest high variability in the types of social sounds used by different humpback whale groups within the same geographic area, and some social sounds overlap with repertoires known from other regions.
Time-domain modelling of underwater transient noise for environmental risk assessment
Authors: Farcas Adrian , Powell Claire F. , Faulkner Rebecca C. , Hastie Gordon D. , Thompson Paul M. , Merchant Nathan D.
Published: November 2016

Underwater transient noise has the potential to cause auditory damage in marine mammals, fish, and invertebrates. The risk of auditory damage is closely linked to the temporal structure of transient signals (e.g., rise time, peak pressure), which is modulated through propagation in the environment. Current models used in environmental risk assessment of underwater noise are based on pulse energy and do not predict the change in temporal structure as the transient propagates away from the source. To address this, the current work presents a time-domain model of transient propagation, based on Fourier synthesis of frequency domain solutions computed using a parabolic equation method. The model outputs are benchmarked first against standard analytical models, then validated against measurements of seismic survey and pile driving noise at various ranges from the source. The results will be discussed in the context of recent regulatory guidelines on the extent of transient acoustic fields and the associated risk of auditory damage to marine mammals.
Lost listening area assessment of anthropogenic sounds in the Chukchi Sea
Authors: Hannay David E. , Matthews Marie-Noel , Schlesinger Angela , Hatch Leila , Harrison Jolie
Published: November 2016

The term listening area, refers to the region of ocean over which sources of sound can be detected by an animal at the center of the space. The lost listening area assessment method has been applied to in-air sounds for a noise effects assessment on birds but not, in our knowledge, previously to the assessment of underwater noise effects on marine mammals. The lost listening area method calculates a fractional reduction in listening area due to the addition of anthropogenic noise to ambient noise. It does not provide absolute areas or volumes of space, as does the communication space method; however, a benefit of the lost listening area method is that it does not rely on source levels of the sounds of interest. Instead, the method depends only on the rate of sound transmission loss. We present a preliminary application of the method from an assessment of “cumulative and chronic effects” of noise produced by oil and gas exploration activities used in the National Marine Fisheries Service's Effects of Oil and Gas Exploration in the Arctic Ocean Final Programmatic Environmental Impact Statement.
Long range sound propagation modeling for assessing seismic survey noise impacts on marine mammals
Authors: Hirai Yukino , Tsuchiya Toshio , Gotoh Shinpei , Shimizu Etsuro , Futa Koji
Published: November 2016

Despite of small land area, Japan has vast EEZ, which is expected to have potential mineral resources. Extensive seismic surveys have been conducted for those targets. However, airguns used for the survey generate intense and low-frequency impulse sound, which could disturb or harm marine mammal behavior and auditory system. In June 2016, NOAA released “Ocean Noise Strategy Roadmap” that recommends modeling of sound propagation in the context of realistic environmental parameters to assess noise impacts on marine creatures. Japan has remarkable seasonal climate changes and complex ocean-bottom topography with trenches and seamounts; therefore, such model is worthwhile. Here, we propose a modeling method utilizing ARGO data includes seasonal climate changes and LEVITUS data, and ocean-bottom topographic configuration data from Google Earth. We calculated long range propagation of airgun sounds in different sound profiles and ocean-bottom topographic configurations around Ogasawara Islands, which is known as a cetacean habitat. Results showed that seasonal climate change and the precision of the ocean-bottom topographic configuration caused significant differences in sound propagation. These findings suggest that the realistic environmental parameters is essentially needed for the modeling of sound propagation and proposed method is a useful tool to assess noise impacts on marine mammals.
Long-term spatially distributed observations of deep diving marine mammals in the Northern Gulf of Mexico using passive acoustic monitoring
Authors: Sidorovskaia Natalia , Li Kun , Tiemann Christopher , Ackleh Azmy , Tang Tingting
Published: November 2016

This paper will present the results of processing long-term passive acoustic monitoring (PAM) data collected July through October 2015 in the Northern Gulf of Mexico in the vicinity of the Deep Water Horizon oil spill site to aid in understanding factors driving the distribution of sperm and beaked whales in the Gulf of Mexico. The Littoral Acoustic Demonstration Center -Gulf Ecological Monitoring and Modeling Consortium (LADC-GEMM) deployed five bottom-anchored acoustic moorings (LADC EARS buoys) at 10, 25, and 50 nmi distance from the 2010 oil spill location. Autonomous surface vehicles and a glider were simultaneously operated in the area for the additional collection of PAM data. The daily and monthly activity of three species of beaked whales exhibits spatial and seasonal variability, which appear to be correlated with levels of anthropogenic noise at the monitoring sites. Acoustic detection data are used to estimate abundances at three sites and compare them to the estimates obtained from baseline data collected before (2007) and right after (2010) the oil spill. Long-term abundance trends for both beaked and sperm whales are discussed. [Research supported by GOMRI.]
Ocean remote sensing with acoustic daylight: Lessons from experiments in the Florida Straits
Authors: Brown Michael G. , Godin Oleg A. , Zang Xiaoqin , Zabotin Nikolay A. , Zabotina Liudmila Y.
Published: November 2016

Ambient and shipping noise in the ocean provides acoustic illumination, which can be used, similarly to daylight in the atmosphere, to characterize the environment. Phase information, which is particularly sensitive to sound speed variations and current velocity, can be retrieved from noise observations by the process known as noise interferometry. Approximations to acoustic Green's functions, which describe sound propagation between two locations, are estimated by cross-correlating time series of diffuse noise measured at those locations. Noise-interferometry-based approximations to Green's functions can be used as the basis for a variety of inversion algorithms, thereby providing a purely passive alternative to active-source ocean acoustic remote sensing. This paper gives an overview of results from noise interferometry experiments conducted in the Florida Straits at 100 m depth in December 2012, and at 600 m depth in September/October 2013. Under good conditions for noise interferometry, estimates of cross-correlation functions are shown to allow one to perform advanced phase-coherent signal processing techniques to: perform waveform inversions; estimate currents by exploiting nonreciprocity; perform time-reversal/back-propagation calculations; and investigate modal dispersion using time-warping techniques. Conditions which are favorable for noise interferometry are identified and discussed. [Work supported by NSF and ONR.]