Recent Developments in Acoustics

Voice Onset Time (VOT) refers to the time difference between two events—one articulatory and the other laryngeal. VOT is measured as the time difference between the onset of the burst (articulatory event) and the onset of vocal fold vibration (laryngeal event) Lisker [14]. The cross-linguistic variation with respect to voicing perception is more in multilingual countries like India. For example, in Tamil language, there is no good distinction between voiced and unvoiced stops; however, in Kannada, Telugu, and Hindi, there exists a difference. To verify the cross-linguistic difference, the behavioral response for VOT continuum in native Kannada and Tamil speakers, and the neuro-physiological changes for VOT continuum in native Kannada and Tamil speakers were taken as objective of the study. Two groups of subjects were participated in the study. Group I consisted of ten male Kannada speakers in the age range of 20–35 years. All the subjects were native speakers of Kannada language, and they belong to Mysore dialect and they are fluent speakers in Kannada only. All though they were exposed to English, they were not fluent speakers of English or Hindi. Group II consisted of ten male Tamil speakers in the age range of 20–35. All the subjects were native speakers of Tamil language, and they belong to Coimbatore/Chennai dialect and they are fluent speakers in Tamil only. Although they were exposed to English, they were not fluent speakers of English or Kannada or Hindi. From the naturally recorded speech sounds, /da-ta/ continuum was created using copy past synthesis method. This continuum had 10 tokens. Using this stimulus, behavioral identification curve was generated. Also electrophysiological, N100 potential was recorded using Neuro-Scan instrument (Compumedies, AUS) with five tokens, which covers the entire dynamic range of the /da-ta/ continuum. The recording was done with 16 channels. The behavioral result showed there was difference between Tamil and Kannada languages; however, electrophysiological results showed that there was a change with N100 latency with changing VOT but there was no language difference found. The author concluded that N100 may not be an effective measure to indicate to represent the voice /voiceless categorical perception. May be higher potential might give better information.

In the present study, a systematic statistical analysis has been performed by the use of words in continuous Garhwali speech corpus. The words of Garhwali in continuous speech corpus are taken from different sources of Garhwali, viz., Newspapers, storybooks, poems, lyrics of songs and magazines, and it showed that there is a quantitative relation between the role of content words in Garhwali and the Shannon information entropy [S] defined by the probability distribution. So far, very few researches have been conducted in Garhwali language. There is no previous knowledge about the syntactic structure of Garhwali language. We have taken finite continuous corpus of Garhwali language. The occurrences of words (frequency) are almost an inverse power law functions, i.e. Zipf’s law and very close to 1.

Formants are the concentration of the acoustic energy around a frequency in the speech wave. Kaur and Narang (1) have noted the variation of pitch and formants in the different age groups in the English language. The current study aimed to find the relation between the aging and formant frequency (F1 and F2) in the Kannada language. The methodology of the study involved eight males and eight females (2 of 8–10 years of age, 2 of 18–20 years of age, 2 of 48–50 years of age, and 2 of above 60 years of age) who were native Kannada speakers. The participants were instructed to repeat the non-sense words V1CV2 in which V2 was /a/always. V1 consisted of three short vowels and three long vowels. The non-sense words used were aka, ika, uka, a: ka, i:ka, and u:ka. The subjects were asked to repeat thrice. In total, 288 tokens were collected for the study. The speech sample is recorded in the Praat software in an acoustically treated room and being the microphone placed 3 cm away from the mouth. The mean of the F1 and F2 of the steady part of the initial vowel (V1) excluding the onset and offset of the three trials is considered. Then, the F1 and F2 were compared across the age group and gender. The present study found that the F1 and F2 decrease as age increases in both males and females.

Objective: The perceptual characteristics of tinnitus are usually assessed by a matching procedure, where loudness and pitch of an external sound are matched to those of the tinnitus percept. For a complete assessment of tinnitus, central auditory processing abilities should be considered in addition to other routine evaluation. Temporal processing is one of the important auditory processing skills that is essential for complex higher level auditory processing. The gap detection test (GDT) and duration discrimination test are relatively simple psychoacoustic methods of measuring temporal resolution [1]. Hence, the present study is aimed at finding the duration discrimination threshold in tinnitus patients at the tinnitus frequency and half an octave above and below the frequency of tinnitus perception. Method: 15 participants with normal or minimal hearing loss with tinnitus in the age range of 18–40 years were enrolled in the present study. Duration discrimination test was administered on all the participants using MATLAB software (MLP toolbox) at the matched tinnitus frequency, half an octave below and above the frequency of tinnitus perception. Results: The results of the study revealed that there was a significant increase in duration discrimination thresholds at the tinnitus frequency compared to half an octave above and below the matched frequency. The result suggests that temporal resolution abilities are affected at the tinnitus frequency in individuals with tinnitus. Conclusions: The results of the study suggest that the temporal processing ability is affected in individuals with tinnitus especially at the frequency of tinnitus.

Professional voice users include singers, teachers, actors, professional speakers, and other entertainers. There are a group of non-elite voice users such as priests, vendors, bus conductors, etc., who are also dependent on their voice for daily living. Temple priests who chant Vedas and mantras the whole day without any form of amplification are at higher risk. Vedas are the foundational literature of Hinduism and it refers to infinitely large collection of mantras. The acoustical characteristics, in addition to the clear phonetic articulation of chants, have deep impact on the devotees. For the renditions to be clear and for it to have an impact, the priests unknowingly tend to put more effort into it. To assess the voice characteristics, perceptual and objective measures have been used extensively. Acoustic analysis is one of the gold standard measures to quantify the voice parameters. The literature mostly concentrates on the vocal load in elite professional voice users. Studies on priests are predominantly in terms of awareness of their voice problems through the use of questionnaires and have majority involved church priests. There are minimal studies done using quantitative acoustic analysis on voice among temple priests. Hence, this study aimed Method: 14 temple priests, in the age range of 35–55 years, who had undergone standard gurukula training, with minimum 5 years of experience were selected. All of them were native speakers of Tamil language. It was made sure that all the participants had only the occupation of being a temple priest, without any other sources of employment, in order to avoid the bias. The participants did the following tasks, namely, reading, chanting, and monologue in their habitual manner. The voice samples were recorded using a digital recorder (Olympus LS) in a quiet environment in the temple. Acoustic analysis was done using real-time pitch software from CSL4500. Results: Findings indicated that the mean frequency range and perturbations were noticeably higher compared to other groups of professional voice users. In general, there was no trend noticed.

Introduction: In children, even with early repair of the cleft, speech errors such as hypernasality, atypical consonant production, and abnormal nasal airflow persist which affects overall speech intelligibility. Perceptual evaluation is considered as the gold standard in the speech assessment of individuals with cleft lip and palate. Speech intelligibility is an important and essential measure of disordered speech; furthermore, it is a major goal of therapeutic intervention. Various factors such as resonance, speech understandability, and speech acceptability vary across stimuli. Aim and Objectives: The present study investigated resonance, speech understandability, and speech acceptability across stimuli (words versus sentences) through perceptual judgment and also correlation among the same was analyzed. Method: The evaluation included 20 native speakers of Kannada language with repaired cleft palate in the age range between 6 and 12 years. The stimuli considered were ten meaningful Kannada words and ten oral Kannada sentences, both loaded with pressure consonants. Responses were collected and were perceptually rated using Henningsson’s rating scale by three experienced speech-language pathologists. Results: The results revealed no significant difference between words and sentences across resonance, speech understandability, and speech acceptability in the paired t-test. Cronbach’s alpha was computed for words and sentences which indicated higher inter-judge reliability among the three judges (α > 0.70). There was a significant correlation (p < 0.05) among resonance, speech understandability, and speech acceptability in both words and in sentences. Conclusion: The present study concludes that resonance, speech understandability, and speech acceptability do not vary across stimuli in children with cleft lip and palate.

Speech recognition is the most emerging field of research, as speech is the natural way of communication. This paper presents the different technologies used for continuous speech recognition. The structure of speech recognition system with different stages is described. Different feature extraction techniques for developing speech recognition system have been studied with merits and demerits. Due to the vital role of language modeling in speech recognition, various aspects of language modeling in speech recognition were presented. Widely used classification techniques for developing speech recognition system were discussed. Importance of speech corpus during the speech recognition process was described. Speech recognition tools for analysis and development purpose were explored. Parameters of speech recognition system testing were discussed. Finally, a comparative study was listed for different technological aspects of speech recognition.

Acoustic analysis is one of the efficient, non-invasive, and quantitative methods of voice assessment. The estimation of glottal flow parameters using acoustic analysis is achieved through the method of Inverse Filtering (IF). This study aims at describing effects of the two available methods of inverse filtering—Formant-Based Inverse Filtering (FBIF) and Linear Prediction-Based Inverse Filtering (LPBIF) on acoustic parameters. The effects of formant deletion and number of LPCs on the various vocal parameters—Fundamental frequency based, intensity based, perturbation based, and noise-based measures—were studied. Phonation samples of /a/ at a comfortable pitch and loudness by 30 healthy participants (15 males and 15 females) were recorded on to a PC in a noise-free environment. In the LPBIF-based method, the recorded voice sample was analyzed under five different conditions, i.e., by varying the number of LPCs. The standard value of LPC used in the Vaghmi software is 18. However, in the current study, the number of LPCs was set to 14, 16, 18, 20, and 22, respectively in each of the analysis conditions with other settings remaining the same. In the FBIF-based method, the recorded voice sample was analyzed under four different conditions, i.e., by deleting different formants from the sample. Formant deletion was accomplished using manual methods. The four conditions used were deletion of F1, deletion of F2, deletion of F3, and deletion of F1, F2, and F3 formants. Acoustic parameters of F0, F0 min, F0 max, I0, I0 min, I0 max, JF0, JT0, RAP3, RAP5, SHdB, APQ5, APQ11, HNR, and GNE were compared in all the different conditions mentioned above. A non-parametric Friedman’s test of differences among repeated measures was conducted. The results revealed no change in the value measured across all conditions in both FBIF and LPBIF methods for the parameters of F0, F0 min, F0 max, I0, I0 min, and I0 max. Significant differences across the four formant conditions under FBIF method were found on all perturbation and noise-related measures, i.e., JF0, JT0, RAP3, RAP5, SHdB, APQ5, APQ11, HNR, and GNE. Significant differences across the five LPC conditions were found only on the parameters SHdB, APQ5, APQ11, HNR and GNE. FBIF could be assumed to be more sensitive than LPBIF as the differences across conditions in FBIF were significant in all measured parameters unlike LPBIF where the differences were significant in only five of the parameters. The pros and cons of using each of the two methods for acoustic analysis of voice are discussed in the current study. Further research needs to be done to investigate the effect of varied parameters of inverse filtering in disordered population.

Introduction: Cleft lip and palate (CLP) is one of the most prevalent congenital deformities seen in Indian children. This condition hinders effective communication in early childhood due to speech and language difficulties. The development of speech and language is mainly based on the type and severity of CLP and the age at which surgical intervention followed by speech therapy was initiated. Early intervention plays an important role in enhancing the communication skills in children with CLP. Objectives: The present study endeavours to compare and profile the speech characteristics of toddlers with repaired cleft of lip and palate (RCLP) with respect to the frequency, type and pattern of speech sound inventory of those who had undergone surgical intervention at an early age versus those who had delayed surgery. Participants: Six Kannada speaking toddlers in the age range of 2–3 years, who were diagnosed to have Expressive Language Delay secondary to repaired cleft of lip and/or palate were considered for the Early Intervention Programme. The six participants were placed into two groups based on the age of surgical intervention. Thus there were three participants in the early intervention group (EIG—who underwent surgery before 1.6 years of age) and three in the delayed intervention group (DIG—who underwent surgery after 1.6 years of age). Method: The baseline was established and measures such as type and pattern of the vowels and consonant inventory were analysed for toddlers in both the groups. Speech and language therapy was given for 20 sessions. Focused stimulation technique was used to enhance speech and language parameters. Post-therapy measurement was done after 20 sessions. Main outcome measures: Descriptive statistics was used to compare the differences in speech measures for the pre-therapy and post-therapy conditions between early and delayed intervention groups. Results: The results revealed certain points of interest. Although both groups showed an improvement in the quality and quantity of phonetic inventory between pre- and post-therapy conditions, it was interesting to note that the mean values in the EI group were consistently higher across phonemes when compared to the DI group. The results clearly highlight the positive effects of early surgical and speech intervention. The speech patterns of each group and the consequent implications have been discussed in detail. Conclusions: The present study is one of the few attempts to investigate the impact of timing of surgical correction on the development of phonetic inventory in children with CLP and also highlights the clinical implications of early surgical and subsequent speech and language intervention.

Sonic crystals are noise barriers which have come to picture since the last two decades for their sound attenuation properties. They consist of sound scatterers arranged periodically inside a host material. The scatterers have high impedance and are put in a fluid of low impedance. Sound attenuation takes place due to multiple scattering of sound waves by the rigid sound scatterers, within particular frequency bands known as Band Gaps. In this paper, a finite element study has been performed on a 2-D sonic crystal having circular scatterers arranged in square pattern. The scatterers are assumed to be sound hard, which imposes that the normal velocity and normal acceleration at their boundary are zero and the arrangement is periodic which is because of the cyclic symmetry of the structure. Relevant boundary conditions have been incorporated into the design which aims in determining the Band Gaps and the corresponding transmission losses through the sonic crystal. Results of eigenfrequency and frequency response analysis of the scatterers are done using a commercial finite element software, COMSOL Multiphysics are presented in this paper.

Acoustic porous materials are extensively used in many engineering applications like building, automobile, aviation, and marine. The health risk factor and environmental claims, associated with traditional materials such as glass wool, mineral fibers, and polymer foams demand for the alternative porous acoustic absorbing materials. Advances in additive manufacturing (AM) allow to manufacture complex structures and give an alternative method to produce porous materials. This study investigates the acoustic properties of porous sound-absorbing material produced by using additive manufacturing (AM) technique and explores the feasibility of AM to manufacture acoustic absorptive materials. For study, three samples with different aperture ratios were fabricated by AM technique, and their sound absorption coefficients were measured experimentally by using the impedance tube. The theoretical formulation for predicting normal sound absorption coefficient of sample with and without air gap was developed and compared with experimental results. The predicted absorption coefficient agrees well with measured results. The measured results indicate that the absorption coefficient of the structures fabricated through AM can be altered by varying aperture ratio and air gap behind the sample. This study reinforces the capability of AM for producing complex acoustic structures with better acoustic properties.

This paper examines the sound transmission loss (STL) through composite panel door systems commonly used in the recording studios, operating theaters, libraries, interview rooms, offices, museums, etc. The sound insulation property of different combinations of door panel of materials such as wood/plywood, cement board, wool, tec sound, laminate is investigated. Reverberation chamber method is used to determine the sound transmission loss. This paper presents the results of a series of doors tests to estimate the sound transmission loss through different combinations of wooden door system. The main aim of this study is to design such a system using wooden door panels that can provide significant noise reduction. The other objective of this study is to evaluate typical doors characteristics including the effect of minor modifications that could increase the noise reduction provided by the door system.

Traditional sound-absorbing materials for noise control are rockwool and glasswool which serve the purpose pretty well but are accompanied by some serious health hazards during their manufacturing or processing or application. Hence researchers and practitioners are coming up with new materials, particularly from natural sources, which apart from being cheap and readily available, have good noise absorption properties along with minimum effect on environment and health hazards. When sound is incident on these materials, some portion is reflected back (due to impedance mismatch), some portion is dissipated within the material (due to thermal and viscous loss in the pores) and the remaining is transmitted to the medium behind the material. Sound absorption is defined as dissipation plus transmission; hence, any parameter which increases any of these two effects will increase the sound absorption. Just like the synthetic sound-absorbing materials, these natural materials are also porous in nature which creates sufficient interest to explore them for potential acoustical applications. Several researchers have conducted acoustical investigations on materials which are used in some other form such as bamboo, jute, kenaf, hemp, ramie, sisal, coconut coir, tea leaf fibre, betel nut fibre and also on some waste materials like recycled rubber, cigarette butt, bagasse, oil palm empty fruit bunch fibre. In this chapter, a mathematical model has been developed to predict the normal sound absorption coefficient of jute and waste cotton. It can also be predicted by estimating the airflow resistivity of a material and feeding that into established models but the amount of error encountered in this approach is often large. So, with an objective to minimise the error, characteristics impedance and complex wave number of the materials have been experimentally estimated. With the minimum sum of squares of error curve fitting approach, a MATLAB code has been developed to obtain the modified coefficients in Delany–Bazley (D–B) model. These equations have been used to predict the sound absorption coefficients of Jute fibres and waste cotton, which is compared with the actual ones obtained by conducting experiments on impedance tube. The modified model was found to be of the greater degree of accuracy than the original D–B model. Further, the airflow resistivity term was eliminated in the Matlab code and the model was used to predict the normal sound absorption coefficient at different frequencies. The results of this model were compared with the result of the original D–B model and experimentally estimated normal absorption. It was also found to be closer to the actual behaviour than the one predicted by D–B model.

Absorption characteristics of fibrous material inherently depend on the microscopic shapes and the dimensions of the fibers. Periodic Unit Cell (PUC) modeling approach is used for the optimization of arrangements of fibers. Periodic hexagonal and square arrangement of fibers are considered here for study. Five Johnson and Champoux-Allard (JCA) parameters and the transmission loss are computed to evaluate the effect of these two structure configurations. Steady Stokes and electric boundary value problem has been solved for estimation of the airflow resistivity, tortuosity and viscous characteristic length, while porosity and thermal characteristic length are estimated directly from mesh geometry. This study underlines the effects due to the change in fiber arrangements on to absorption characteristics of 50 mm thick sample size of the rigid porous medium. It is observed that for equal centric diameter of fibers hexagonal configuration yields better transmission loss compared to square configuration of fibers over frequency range of 0–8 kHz.

Srivastava shown in Srivastava (Powder Technol 320:108–113, 2017, Lect Notes Mech Eng 19:149–155) that the velocity of the steady flow outside the boundary layer was independent of viscosity but was dependent on permeability. Through this paper, we tried presenting practical implication of the theory developed by Srivastava (Powder Technol 320:108–113, 2017, Lect Notes Mech Eng 19:149–155). It was found that the flow may be described using the non-dimensional parameters. Results were plotted for the volumetric flow rate analysis of the problem.

A fan rotor system can generate a large amplitude blade-passing frequency (BPF) component if the gap between rotating fans and stationary diffusers is not equal. A series of experiments was conducted in this study using Spectra Quest’s Machinery Fault Simulator (MFS) to observe the behavioral changes of the BPF and its harmonics before and after installing an obstruction nearby an axial fan. In order to incorporate the flexibility of the shaft a modification in the existing setup (MFS-LITE) has been designed and fabricated. The whole study is divided into two parts: in the first part readings were taken without considering the effect of flexibility of shaft and for performing this, the shaft of smaller length is taken; and in the second part readings were taken with a longer shaft in order to incorporate the effect of flexibility during experimentation. The result shows that the obstruction had an obvious and consistent effect on the 1BPF and 2BPF components for the axial fan rotor system.

There are ample literature addressing the dynamic analysis for gears. In this paper, some important literature on dynamics of both healthy gears and gears with defects have been reviewed. Dynamic analysis of gears is significant from the condition monitoring point of view. Single degree of freedom dynamic model for spur gears has been detailed first. Dynamic study for worm gear with force analysis has been reported. Also, the effect of defects like crack and spall on dynamic behavior of gears studied by researchers has been included in this review. The study of defect model can be utilized in improving the efficiency of the system and preventing the failures in industries as well.

The Propeller is a vital component which is delicate for the safe operation of a ship at sea. The propeller noise is very much significant to warship designers and military strategists for many years. By using large eddy simulation (LES) it is possible to simulate the sound radiation of turbulent flows effectively, providing access to resolve turbulent scales at higher accuracy. The main aim of this paper is to study and identify a propeller which is producing less noise. The present paper deals with the prediction of the unsteady non-cavitating marine propeller noise of 5, 6, and 7 blades with +10° pitch, +5° pitch, existing pitch, and −5° pitch, −10° pitch at rotational speed 840 rpm and vehicle speeds of 7.62 m/s, using eddy viscosity model of large eddy simulation (LES) available in computational fluid dynamics fluent software and using Ffowcs Williams–Hawkings (FW-H) formulation. Here solver is chosen as pressure-based, unsteady formulation of second-order implicit. Finite volume method is used to predict the noise in time-domain acoustic analogy. Sound pressure levels are predicted at different receiver positions. The receiver position is 1 m distance in radial direction to the propeller. From this numerical study on these propellers, it is found that the propeller of 6 blades with +5° pitch generates least noise. Finalized the propeller which is producing low noise as well as which generates required Thrust and Torque by changing the diameter of the propeller. Finally, obtain the noise reduction of 9.4 dB with the new redesigned propeller when compared with the baseline propeller for the same thrust and torque.

Any underwater vehicle like submarine, torpedo and AUV uses a propeller for propulsion. The significance of propeller is to develop thrust which propels the vehicle at its operational speed. The propeller geometry and design are complicated which involves many controlling parameters. The pressure difference is produced on the propeller blade between the forward and rear surfaces which accelerates water behind the blade. In addition to thrust generation, it creates noise which could be detected by the enemy. From stealth point of view, this noise is to be reduced. The present work is aimed at controlling the noise generated by the propeller by changing the skew angle of the propeller. The aim of this paper is to identify the skew angle of a propeller at which propeller noise is least. Unsteady non-cavitating noise of marine propeller of six blades with +5^° skew, +10^° skew, +15^° skew and +20^° skew and existing propeller at rotating speed of 780 rpm and vehicle speed of 7.08 m/s was predicted. The methodology adopted in CFD analysis is eddy viscosity model of Large Eddy Simulation (LES) and for acoustic analysis is the Ffowcs Williams–Hawkings (FW-H) formulation. From this numerical study on these propellers, it is found that the propeller having +15^° skew angle generates the least noise.

The Aeolian tones radiated from structures are important in various engineering fields. For example, in military and defense applications, it is used for detection and survivability. Aeroacoustic noise is increasingly used as a critical design variable in modern engineering designs. This in turn has led to an increase in the research efforts aimed at numerical prediction of aerodynamic noise, often dubbed computational aeroacoustics (CAA). A numerical study of aerodynamically generated acoustic noise was carried out. Simulations of 2D, laminar, and compressible, viscous flow were performed at Reynolds Number (Re) ∈ [60, 200] and Mach no. (Ma) ∈ [0.2, 0.4]. Roe Flux-Difference Splitting Scheme was used to solve the governing equations with finite volume differencing. The Least square cell based spatial discretization scheme was used with second-order upwind scheme for calculating convection terms. Acoustic calculations were done using Ffowcs-Williams and Hawkings (FW-H) formulation. The peak in the receiver spectra matches the dominant frequency in the lift force history. Sound Pressure level decreases with decreasing flow velocity and Reynolds number. Acoustic peak frequency shifts toward higher frequency value with increasing Mach number for all cases of Reynolds number. The sound pressure level was observed to be higher downstream of the flow. Wide band noises at frequencies higher than the fundamental frequency were observed to be the subharmonics of the fundamental frequency of sound.

The direction of arrival (DoA) of an acoustic source is a salient parameter whose applications are diversified ranging from the defense industry to the entertainment industry. The conventional method of DoA estimation uses sensor (microphone or hydrophone) array with different computationally intensive algorithms, however, the size of sensor-array restricts its deployment in portable devices. The replacement of sensor-array is an acoustic vector sensor (AVS), which is compact in size and can be easily mounted on any small portable device. The AVS can be realized with omnidirectional microphones and/or particle velocity sensors. We have used different geometrical arrangements of closely spaced omnidirectional microphones (called AVS configurations) for determining the DoA of a speech source for real-time applications. The signals acquired by the closely spaced omnidirectional microphones are used to estimate the acoustic intensity vector, which gives the direction of an acoustic source. In this paper, we have given a method for DoA estimation of a broadband quasi-stationary source (speech signal) and compared the DoA estimation performance of different omnidirectional microphones based AVS configurations with the help of Finite Element Method tool, viz., COMSOL Multiphysics. The speech signals for different angular locations at different microphones of an AVS have been recorded using COMSOL Multiphysics modeling software.

The vibration signals generated by a rotor bearing system are greatly influenced by the occurrence of a fault on the bearings. Monitoring of parameters like peak, overall rms, kurtosis, crest factor, and power from the time domain signal reveals the status of health of bearings supporting the rotor and serves as the easiest approach. However, the majority of these parameters give an estimation of the overall health of the system and not just the bearings. This work addresses the use of Discrete Wavelet Transform (DWT) to filter the bearing related signal and then monitor the health of bearing. The work also discusses on the selection of appropriate sampling frequency for the collection of time domain signal and demonstrates the dependence of parameters on the same. The results indicate that the use of DWT for a signal with higher sampling frequency has increased the accuracy of prediction of estimation of the defect.

Acoustic Emission Technique (AET) is a state-of-the-art method to measure faint surface vibration resulting from stress waves when a material is subjected to crack, fracture, corrosion, friction, wear, turbulence, etc. Acoustic emission is a widely used and standardized technique in the field of structural diagnosis. The technique has been extended to machinery condition monitoring in recent years. However, with the non-availability of standards dedicated to the machinery section, the measurements are mostly carried out on personal experiences. This article is a set of standard practices, drawn from already established standards, new methods conceived and applied by researchers in their work and authors’ personal experiences, for application of AET in rotating machinery.

We characterize the timbre of the Sarasvati Veena, a traditional Indian string instrument, by analysing the sound spectrum and the energy distribution with different frequencies as a function of time. We compare the acoustic analysis with that of a simple sonometer. We find that while the spectrum consists of the fundamental and harmonics of a plucked string under tension, the Veena shows a prominent formant near the resonant frequency of the body. But more notable is the behaviour of the higher harmonics (upper partials) which share substantial energy and show a characteristic revival with time, in contrast to the sonometer spectrum.

With the advent of various techniques to assess the bioelectric signals on the surface of the body, it has become possible to develop various Human–Computer interface systems. In this study, for the first time a cross-correlation based data is reported for two different types of bio-signals, viz. Electroencephalography (EEG) and Electromyography (EMG). Whereas EEG refers to the neuro-electric impulses generated in the brain recorded in the form of electric potentials, EMG records the activation potentials of the muscle cells when they contract or relax. The ability of Hindustani Music (HM) to evoke a wide range of emotional experience in its listeners is widely known. For this study, we took simultaneous EEG and EMG data of 5 participants while they listened to two Hindustani ragas of contrast emotions namely Chayanat (corresponding to happy/joy) and Darbari Kanada (corresponding to sad/pathos) emotion. We make use of two latest signal processing algorithms—Wavelet-based power spectra and cross-correlation coefficient to assess the arousal based activities in response to the acoustic clips in the two different bio-signals. For the first time, an attempt is being made to quantify and categorize musical emotions using EMG signals and an attempt to correlate that with the EEG signals obtained from the brain. The alpha, theta, and gamma frequency range in the frontal and parietal electrodes is found to be the most responsive in case of musical emotions. The EMG response has been studied by segregating the entire signal into different frequency ranges as is done in case of EEG frequency bands. Interestingly, the response in case of EMG data is strongest in the same frequency bands as that of EEG signals. Novel pitch detection algorithm has also been applied to EMG signals to ratify the rationale behind the separation of frequency bands. This is the first of its kind study which looks for categorization and quantification of musical emotions using simultaneously two different bio-signals with the help of robust mathematical analysis. The results and implications have been discussed in detail.

The Atmospheric Boundary Layer (ABL) was continuously monitored using Monostatic SODAR, which was conducted at CSIR-NPL, New Delhi. In this paper, an effort has been made to review the studies in the field of the structure of ABL using SODAR echograms in Delhi during a special observation period of the experiment. The change in weather conditions has found to have definite marks on the SODAR observed the structure of ABL. In particular, it has been seen that dot echoes and spiky top layered structures are different typical structures of the different seasons, which are associated with temperature, wind speed, and relative humidity. Also, the knowledge of these structures is great to use in air pollutant studies and dispersion models.