Theoretical and phantom based investigation of the impact of sound speed and backscatter variations on attenuation slope estimation

Quantitative ultrasound features such as the attenuation slope, sound speed and scatterer size, have been utilized to evaluate pathological variations in soft tissues such as the liver and breast. However, the impact of variations in the sound speed and backscatter due to underlying fat content or fibrotic changes, on the attenuation slope has not been addressed. Both numerical and acoustically uniform tissue-mimicking experimental phantoms are used to demonstrate the impact of sound speed variations on attenuation slope using clinical real-time ultrasound scanners equipped with linear array transducers. Radiofrequency data at center frequencies of 4 and 5 MHz are acquired for the experimental and numerical phantoms respectively. Numerical phantom sound speeds between 1480 and 1600 m/s in increments of 20 m/s for attenuation coefficients of 0.3, 0.4, 0.5, 0.6, and 0.7 dB/cm/MHz are simulated. Variations in the attenuation slope when the backscatter intensity of the sample is equal, 3 dB higher, and 3 dB lower than the reference is also evaluated. The sound speed for the experimental tissue-mimicking phantoms were 1500, 1540, 1560 and 1580 m/s respectively, with an attenuation coefficient of 0.5 dB/cm/MHz. Radiofrequency data is processed using three different attenuation estimation algorithms, i.e. the reference phantom, centroid downshift, and a hybrid method. In both numerical and experimental phantoms our results indicate a bias in attenuation slope estimates when the reference phantom sound speed is higher (overestimation) or lower (underestimation) than that of the sample. This bias is introduced via a small spectral shift in the normalized power spectra of the reference and sample with different sound speeds. The hybrid method provides the best estimation performance, especially for sample attenuation coefficient values lower than that of the reference phantom. The performance of all the methods deteriorates when the attenuation coefficient of the reference phantom is lower than that of the sample. In addition, the hybrid method is the least sensitive to sample backscatter intensity variations.     

A study of the accuracy of mobile technology for measuring urban noise pollution in large scale participatory sensing campaigns

The study reports on the relevancy and accuracy of using mobile phones in participatory noise pollution monitoring studies in an urban context. During one year, 60 participants used the same smartphone model to measure environmental noise at 28 different locations in Paris. All measurements were performed with the same calibrated application. The sound pressure level was recorded from the microphone every second during a 10-min period. The participants frequently measured the evolution of the sound level near two standard monitoring sound stations (in a square and near a boulevard), which enables the assessment of the accuracy and relevancy of collected acoustic measurements. The instantaneous A-weighting sound level, energy indicators such as L_A,eq, L_A10, L_A50 or L_A90 and event indicators such as the number of noise events exceeding a certain threshold L_α (NNEL ? L_α) were measured and compared with reference measurements. The results show that instantaneous sound levels measured with mobile phones correlate very well (r > 0.9, p < 0.05) with sound levels measured with a class 1 reference sound level meter with a root mean square error smaller than 3 dB(A). About 10% of the measurements for the boulevard location (respectively 20% for the square) were inaccurate (r < 0.3, p < 0.05). Nevertheless, mobile phone measurements are in agreement for the L_A50 and the L_A90 acoustic indicators with the fixed station (4-m high) measurements, with a median deviation smaller than 1.5 dB(A) for the boulevard (respectively 3 dB(A) for the square).     

Comparison between measured and calculated parameters for the acoustical characterization of small classrooms

This paper presents a comparison between measured and calculated acoustical parameters in eight high school classrooms. The mid frequency unoccupied and occupied reverberation times and the 1 kHz sound propagation (SP) of the reverberant and total speech levels in occupied classrooms were compared with analytical and numerical predictions. The ODEON 6.5 code and the Sabine formula gave the most accurate results for reverberation time in the empty classrooms with overall relative differences of 8.1% and 9.7%, respectively. With students present, the Eyring and Sabine formulas and Hodgson’s empirical model resulted to be the most accurate with relative differences of 11.1%, 13.2% and 13.6%, respectively. The reverberant speech levels decrease with increasing distance from the source at rates varying from −1.21 to −2.62 dB/distance doubling, and the Hodgson model fits the slope values quite well. The best predictions of the SP of the reverberant and total speech levels are shown, in order of accuracy, for the ODEON code, the Barron and Lee theory and the classical diffuse field theory. Lower rms errors were found when the measured total acoustic absorptions were used. The lowest rms error of 1.4 dB for the SP of the total speech level were found for both the ODEON code and the Barron and Lee theory.     

Development of a multi-microphone calibrator

This paper presents the theory, design, and validation of a microphone calibrator used to simultaneously calibrate the amplitudes of multiple microphones on a single probe. The probe uses four 6 mm diameter electret microphones to acquire the data needed to compute acoustic energy density. This probe has prompted the need for simultaneous, multi-microphone amplitude calibration. The calibration process simultaneously subject each microphone on the probe to the same known acoustic pressure using four equal-length, small-diameter tubes connected to a single excitation source. A reference microphone connected to a fifth tube is used to calibrate the microphones. Test results show that the calibrator can calibrate each probe microphone within ±0.5 dB up to 2 kHz, and within ±1 dB up to 4.9 Hz with a confidence level of 95%.     

Determination of the sound energy level of a gunshot and its applications in room acoustics

In some cases an impulsive noise source such as a gunshot can be a preferred alternative when investigating building acoustics, including sound insulation measurements, when compared to conventional steady state noise sources. A gun equipped with blank cartridges is an impulsive noise source that is lightweight and small enough to be easily transported. The differences in the noise characteristics between individual cartridges for the same gun are usually small, so the impulsive source can be replicated to a high degree. This paper is focused on the practical application of the sound exposure levels produced by a gunshot with a known sound energy level in the rooms under investigation. In this way, the equipment and methods required by the conventional method are simplified significantly. Furthermore, reverberation times need not be measured, since the equivalent absorption area can be directly obtained from the measured sound exposure levels. Using Green’s theorem, the roles of the sound source and measuring microphone were exchanged, which simplified the determination of sound insulation as it was easier to change the position of the gun than the microphone. The results obtained using the impulsive noise source were in good agreement with those obtained using the conventional method. Above 100 Hz, their difference in any frequency band of interest was less than 1 dB.     

Vocal effort with changing talker-to-listener distance in different acoustic environments

Talkers adjust their vocal effort to communicate at different distances, aiming to compensate for the sound propagation losses. The present paper studies the influence of four acoustically different rooms on the speech produced by 13 male talkers addressing a listener at four distances. Talkers raised their vocal intensity by between 1.3 and 2.2 dB per double distance to the listener and lowered it as a linear function of the quantity "room gain" at a rate of -3.6 dB/dB. There were also significant variations in the mean fundamental frequency, both across distance (3.8 Hz per double distance) and among environments (4.3 Hz), and in the long-term standard deviation of the fundamental frequency among rooms (4 Hz). In the most uncomfortable rooms to speak in, talkers prolonged the voiced segments of the speech they produced, either as a side-effect of increased vocal intensity or in order to compensate for a decrease in speech intelligibility.     

Verbal communication and noise in eating establishments

A new simple prediction model has been derived for the average A-weighted noise level due to many people speaking in a room with assumed diffuse sound field. Due to the feed-back influence of noise on the speech level (the Lombard effect), the speech level increases in noisy environments, and the suggested prediction model gives a 6 dB reduction of the noise level by doubling the equivalent absorption area of the room. This is in contrast to the lowering by 3 dB by doubling of the absorption area for a constant power sound source. The prediction model is verified by experimental data found in the literature. In order to achieve acceptable conditions for speech communication within a small group of people, a guide for the recommended minimum absorption area per person in eating establishments is provided.     

Coded excitation for ultrasound tissue harmonic imaging

Coded excitation can improve the signal-to-noise ratio (SNR) in ultrasound tissue harmonic imaging (THI). However, it could suffer from the increased sidelobe artifact caused by incomplete pulse compression due to the spectral overlap between the fundamental and harmonic components of ultrasound signal after nonlinear propagation in tissues. In this paper, three coded tissue harmonic imaging (CTHI) techniques based on bandpass filtering, power modulation and pulse inversion (i.e., CTHI-BF, CTHI-PM, and CTHI-PI) were evaluated by measuring the peak range sidelobe level (PRSL) with varying frequency bandwidths. From simulation and in vitro studies, the CTHI-PI outperforms the CTHI-BF and CTHI-PM methods in terms of the PRSL, e.g., −43.5 dB vs. −24.8 dB and −23.0 dB, respectively.     

Effect of vocal effort on spectral properties of vowels.

The effects of variations in vocal effort corresponding to common conversation situations on spectral properties of vowels were investigated. A database in which three degrees of vocal effort were suggested to the speakers by varying the distance to their interlocutor in three steps (close--0.4 m, normal--1.5 m, and far--6 m) was recorded. The speech materials consisted of isolated French vowels, uttered by ten naive speakers in a quiet furnished room. Manual measurements of fundamental frequency F0, frequencies, and amplitudes of the first three formants (F1, F2, F3, A1, A2, and A3), and on total amplitude were carried out. The speech materials were perceptually validated in three respects: identity of the vowel, gender of the speaker, and vocal effort. Results indicated that the speech materials were appropriate for the study. Acoustic analysis showed that F0 and F1 were highly correlated with vocal effort and varied at rates close to 5 Hz/dB for F0 and 3.5 Hz/dB for F1. Statistically F2 and F3 did not vary significantly with vocal effort. Formant amplitudes A1, A2, and A3 increased significantly; The amplitudes in the high-frequency range increased more than those in the lower part of the spectrum, revealing a change in spectral tilt. On the average, when the overall amplitude is increased by 10 dB, A1, A2, and A3 are increased by 11, 12.4, and 13 dB, respectively. Using "auditory" dimensions, such as the F1-F0 difference, and a "spectral center of gravity" between adjacent formants for representing vowel features did not reveal a better constancy of these parameters with respect to the variations of vocal effort and speaker. Thus a global view is evoked, in which all of the aspects of the signal should be processed simultaneously.     

Experimental and theoretical studies on a double-pass C-band bismuth-based erbium-doped fiber amplifier

Performance of a Bismuth-based Erbium-doped fiber amplifier is experimentally and theoretically investigated using 1480 nm pumping with double-pass scheme. In the theoretical analysis, the rate and power propagation equations are solved to examine the optimum length for the C-band operation as well as the gain and noise figure characteristics. The calculated small signal gain is 38 dB with gain variation of less than 3 dB. The measured gain is 4 dB lower due to spurious reflections which were ignored in the theoretical analysis. At input signal power of 0 dBm, a gain of 14.5 dB is obtained experimentally with gain variation of less than 1 dB within the wavelength region from 1530 to 1565 nm. The noise figure is less than 12 dB within this region.     

Acoustic methods for measuring bullet velocity

This article describes two acoustic methods to measure bullet velocity with an accuracy of 1% or better. In one method, a microphone is placed within 0.1 m of the gun muzzle and a bullet is fired at a steel target 45 m away. The bullet’s flight time is the recorded time between the muzzle blast and sound of hitting the target minus the time for the sound to return from the target to the microphone. In the other method, the microphone is placed equidistant from both the gun muzzle and the steel target 91 m away. The time of flight is the recorded time between the muzzle blast and the sound of the bullet hitting the target. In both cases, the average bullet velocity is simply the flight distance divided by the flight time.     

Factors affecting sound exposure from firing an SA80 high-velocity rifle

The effect of distance on the peak sound pressure level and sound exposure level from an SA80 rifle has been investigated. Sound pressure waveforms were measured in two directions from the gun: downrange, from 50 m to 300 m, and to the left-hand side, from 0.3 m to 32 m. Some additional measurements were made to the right of the gun. Measurements made downrange showed three distinct features of the waveform; the shock wave from the supersonic bullet, the reflection from the ground, and the muzzle blast. The time elapsed between the shock wave and the muzzle blast increased with increasing distance: 94 ms for a distance of 50 m, and 507 ms for a distance of 300 m. The highest peak sound level downrange from a single round was between 151 dB(C) and 148 dB(C) at distances from 50 m to 300 m, and varied little if at all with distance. To the left of the gun, the peak sound pressure level of 161 dB(C) at 0.3 m reduced to 128 dB(C) at 32 m. The peak sound pressure level was estimated to be 137 dB(C) at a distance of approximately 20 m to the left-hand side. Hearing protection must therefore be worn by anyone closer than 20 m to a person firing. The peak sound pressure level was estimated to be 135 dB(C) at a distance of approximately 25 m and therefore hearing protection is recommended at distances of up to 25 m. The sound exposure level of 98 dB(A) at 20 m indicated that an observer at this distance could hear about 1440 rounds without hearing protection before the noise exposure reached the upper exposure action value specified in the Control of Noise at Work Regulations 2005. Peak sound pressure levels were on average 2.4 dB higher at the left ear compared with the right ear.     

A possible acoustic design approach for multi-purpose auditoria suitable for both speech and music

In several auditoria, it has been observed that the reverberation time is longer than expected and that the cause is a horizontal reverberant field established in the region near the ceiling, a field which is remote from the sound absorbing audience. This has been observed in the Boston Symphony Hall, Massachusetts, and the Stadthalle Göttingen, Germany. Subjective remarks on their acoustics suggest that there are no unfavourable comments linked to the secondary sound field. Two acoustic scale models are considered here. In a generic rectangular concert hall model, the walls and ceiling contained openings in which either plane or scattering panels could be placed. With plane panels, the model reverberation time (RT) was measured as 53% higher than the Sabine prediction (frequency 500/1000 Hz), compared with 8% higher with scattering panels. The second model of a 300 seat lecture theatre with a 6 m or 8 m high ceiling had raked seating. In this case, the amount of absorption in the model was increased until the point was reached where speech had acceptable intelligibility, with the early energy fraction, D ? 0.5. For this acceptable speech condition with the 6 m ceiling, the measured mid-frequency T₁₅ was 1.47 s, whereas the Sabine predicted RT was 1.06 s. The sound decay was basically non-linear with T₃₀ > T₁₅ > EDT. Exploiting a high-level horizontal reverberant field offers the possibility of acoustics that are better adapted as suitable for both speech and unamplified music, without any physical change in the auditorium. Using secondary reverberation in an auditorium for a wide variety of music might also be beneficial.     

Compact SOI arrayed waveguide grating demultiplexer with broad spectral response

A compact eight-channel flat spectral response arrayed waveguide grating (AWG) multiplexer based on silicon-on-insulator (SOI) materials has been fabricated on the planar lightwave circuit (PLC). The 1-dB bandwidth of 48 GHz and 3-dB bandwidth of 69 GHz are obtained for the 100 GHz channel spacing. Not only non-adjacent crosstalk but also adjacent crosstalk are less than −25 dB. The on-chip propagation loss range is from 3.5 to 3.9 dB, and the total device size is 1.5 × 1.0 cm².     

Excess attenuation for sound propagation over an urban canyon

Because quiet areas in dense urban environments are important to well-being, the prediction of sound propagation to shielded urban areas is an ongoing research focus. Sound levels in shielded areas, such as canyons between rows of buildings, are strongly influenced by distant sources. Therefore, propagation factors such as metrology, screening, and intermediate canyons—as occur between a source canyon and a receiver canyon—must be addressed in an engineering propagation model. Though current models address many important propagation factors, engineering treatment of a closed urban canyon, subject to multiple internal reflections, remains difficult.A numerical investigation of sound propagation across the open tops of intermediate urban canyons has been performed, using the parabolic equation and equivalent sources methods. Results have been collected for various canyon geometries, and the influences of multiple canyons, canyon/rooftop absorption, variable rooftop height, wind gradient, and correlated versus uncorrelated source models have been investigated. Resulting wideband excess attenuation values ranged from −1 dB to −4 dB per canyon, and were fairly constant with frequency in many useful cases. By characterizing the excess attenuation of canyons intermediate to the source and receiver, the influence of these intermediate canyons could be addressed simply, without the overhead of a detailed numerical calculation.     

Cascadability and reshaping properties of a saturable absorber inserted inside a RZ transmission line for future 160-Gbit/s all-optical 2R-regenerators

In this prospective work, we analyze the behavior of a quantum-well microcavity saturable absorber component cascaded into a 100-km SMF RZ transmission line in order to annihilate the ghost-pulse phenomenon in the following simplified “…010101…” 160-Gbit/s 2-bit pattern at 1555 nm. Recirculating-loop experiments show a maximal ghost-pulse extinction up to 11.6 dB as well as an intensity extinction ratio enhancement higher than 6 dB over at least 800 km of propagation.     

Real-time aircraft noise likeness detector

One of the most difficult tasks involved in the process of noise monitoring near airports is related to the automatic detection and classification of aircraft noise events.These tasks can be solved by applying pattern recognition techniques to the audio signal captured by a microphone. But now the problem is caused by the background noise, which is present in real environments.This paper proposes a real-time method for continuously tracking the similarity of the input sound and the aircraft’s sounds. Using these facilities, the monitoring unit will be able to mark aircraft events, or to make measurements only when aircraft sound is louder than background noise.A one-class approach has been applied to this detection-by-classification method.Using the default setup, 93% of the aircraft’s events which held an SNR of 6-8 dB were detected, for 30 different locations with diverse soundscapes.     

Speech transmission performance and the effect of acoustical remedies in a dome

For the purpose of improving speech transmission performance in a dome space, the acoustical properties in a dome having a diameter of 20 m were examined. The acoustical properties measured evenly on the floor of the dome were evaluated both objectively and subjectively and the interrelationship of the objective measures and subjective measures were also examined. Then, on the basis of the results of the study, simplified acoustical remedies were applied to the dome to improve speech intelligibility and the effect of the remedies was also examined. The following findings were obtained from this investigation.(1) The speech transmission performance in the dome space without treatment by absorptive materials varies greatly with the locations of sound sources and observation points: a range of 0.17-0.59 for RASTI value and a range of 30-97% for speech intelligibility test results. (2) There are peculiar observation points at which speech transmission quality is very high due to a considerable sum of the energy arriving in the first 0.06 s after the arrival of the direct sound. (3) Of all the measured acoustical parameters, RASTI, EDT in 1 kHz band, early-to-late arriving sound energy ratio, and Ts corresponded well to the speech intelligibility test scores. (4) Rubber tiles, cotton canvas 12 m in length, and glass wool board, are effective in improving speech intelligibility remarkably due to increased sound absorption and the diffusion effect.     

Skirts and barriers for reduction of wayside noise from railway vehicles—an experimental investigation with application to the BR185 locomotive

An experimental investigation to determine the noise reduction efficiency of a number of combinations of vehicle mounted noise skirts and trackside low barriers has been carried out. A 1:4-scale mock-up of the German BR185 locomotive was built. Special care was taken to achieve a realistic representation of the wheel/rail sources, using rotating acoustic source wheels able to imitate the radiation from structural modes and acoustic rail ducts with independently adjustable vertical and lateral slots. The acoustic insertion loss (IL) equivalent to a full-scale microphone position at 25 m distance from the track was determined for the different source components separately. The total IL was obtained from sound power spectra calculated with the TWINS software. Results for the design speed (v=120 km/h) and a case with a lower speed (v=100 km/h) are presented to illustrate the effect of speed on the acoustic IL. The tests were performed in open-air free field conditions. The experimental procedure used in the present investigation gives detailed information on the relative contributions from different source components, which is valuable for further design studies. For the eight combinations reported here, the overall reduction achieved was in agreement with results in the literature. The IL was 2-3 dB(A) for cases with only vehicle skirts and the case with only low track barriers. The combined configurations had insertion losses of 7-13 dB(A).