A Robust Multichannel Lung Sound Recording Device
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This paper presents a robust multichannel lung sound recording device (LSRD) for automatic lung sound classification. Compared to common approaches, we improved the usability and the robustness against body sounds and ambient noise. We developed a novel lung sound transducer (LST) and an appropriate attachment method realized as a foam pad. For analogue prefiltering, preamplification, and digitization of the lung sound signal, we use a composition of low-cost standard audio recording equipment. Furthermore, we developed a suitable recording software. In our experiments, we show the robustness of our LSRD against ambient noise, and we demonstrate the achieved signal quality. The LSTâs microphone features a signal-to-noise ratio of SNR = 80 dB. Therefore, we obtain a bandwidth of up to a frequency of f â 2500 Hz for vesicular lung sound recordings. Compared to the attachment of the LST with self-adhesive tape, the foam pad achieves an attenuation of ambient noise of up to 50 dB in the relevant frequency range. The result of this work is a multichannel recording device, which enables a fast gathering of valuable lung sounds in noisy clinical environments without impeding the daily routines.Keywords:
Robustness
Ambient noise level
Sound Quality
Taken sound fullness,brightness,definition,softness,strength and balance as evaluation index of sound quality,sound quality was evaluated and analyzed for a group of homemade specialized microphone.Based on its subjective testing,sound quality of homemade microphone was analyzed and compared with that of foreign products.
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Tentative damage-risk criteria for hearing and material design standards require measuring the peak values of impulsive sounds. However, use of current or proposed ANSI and IEC standards provides only the magnitude calibration in response to sinusoids, and does not ensure accuracy in measuring the peak values of impulsive sounds. We have used digital transient capture and FFT techniques with a 3.2-mm-diam microphone system calibrated in magnitude and phase to measure peak sound pressure levels and spectra of impact sounds produced by a prototype source. This source uses an essentially identical pair of colliding steel spheres of either 25.4- or 44.5-mm diameter to produce peak sound pressure levels within the range 100-120 dB re 0.00002 Pa with precisions (ranges of peak amplitudes of nominally identical impacts) of 0.5 dB or better, and with spectra exhibiting fairly broad maxima within the range 3-10 kHz. For some conditions, at least, the waveform of sound pressure is approximately diphasic, and of duration <0.5 ms. The results demonstrate the feasibility of a simple, robust sound source suitable for use in moderately noisy, nonanechoic environments to check the free-field response of microphone systems, and subsequent instrumentation, to impulsive sounds.
Instrumentation
Transient (computer programming)
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A procedure for sound filed simulation, sound quality (SQ) evaluation, and optimization of interior noise of a rail vehicle is investigated in this paper. Firstly, some interior noises are measured on site when the subway is running in tunnel at a speed of 60 km/h. The sound pressure levels (SPLs), loudness, sharpness, and roughness of the measured noise are analyzed. A finite element model for acoustical simulation of the carriage is established by using the Actran software. The accuracy and feasibility of the finite model are verified by comparing the psychoacoustical parameters from the simulations and measurements. By using orthogonal experimental design, finally, the best optimization scheme is put forward, which obtained a sound quality improvement with a 4.81 dB decrease in SPL and a 1.07 sone reduction in loudness. The proposed optimization scheme may be extended to other vehicles for improving interior acoustic environment.
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The spatial correlation of the sound pressure and the particle velocity in surface-generated noise was investigated.Method of calculating the correlation between the sound pressure and the particle velocity in ambient noise was developed based on ray treatment.The correlation of the sound pressure and the particle velocity in ambient noise model with given surface and bottom power reflection coefficients and volume absorption could easily be got by numerical integration.Results of given condition were shown.
Hydrophone
Ambient noise level
Particle velocity
Reflection
Spatial correlation
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The directivity of a microphone is the relative sensitivity variation as a function of incidence angle. This is an important characteristic when every sound arriving at the microphone position is relevant. This paper presents the measuring directivity of half-inch measurement microphones. The microphone is placed in a free sound field and exposed to a sound pressure. The unloaded output voltage is measured. Then, the microphone is rotated in 30o steps, while both the acoustic centers of the microphone and the sound source remain in fixed positions in the sound field, and is exposed to the same sound pressure. The unloaded output voltage for each angle of incidence is measured and them the difference relative to 0 deg. of incidence is calculated as a function of frequency. The directivity of eight measurement microphones from the same manufacturer but of different types and frequency response characteristics (i.e. pressure field, random incidence or free-field) were measured. Results showed that the microphones object of this investigation have similar directivities in the frequency range from 16 Hz to 12.5 kHz regardless of their frequency response characteristics.
Directivity
Free field
Noise-canceling microphone
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Probe microphone measurement systems allow determination of sound pressure level (SPL) within the external auditory meatus (EAM). EAM geometry and middle ear impedance contribute significantly to the acoustic properties of the EAM, and consequently sound pressure distribution along the EAM is not uniform. Accurate measurement of SPL at the tympanic membrane (TM) requires probe microphone placement within 6 to 8 mm of the TM. A method of probe placement using the acoustical properties of the EAM (Sullivan, 1988) was investigated in 6 adults. Sullivan's acoustic probe placement method was found to overestimate probe distance from the TM, but with minor modification the acoustic method could be used to place the probe so that real ear measurements accurately predicted SPL at the TM. The difference between TM SPL and SPL at probe position was compared for the acoustic method and a constant insertion depth (25 mm) method. More accurate estimates of TM SPL were obtained with the acoustic method.
Acoustic impedance
Meatus
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Sound Quality
Annoyance
Rectangle
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Artificial mouths used in telephonometric measurements are calibrated to produce a frequency-independent sound pressure of a known level at a given calibration point. Common practice in the telephone industry has been to measure this sound pressure with a Type L Laboratory Standard Microphone, using its free-field response curve. For a fixed artificial mouth calibration, the sound pressure indicated by various Type L microphones has been determined over the 100–10000-Hz frequency range. The results show an expected difference at high frequencies between “free-field” microphones and “pressure” microphones with a free-field correction applied. They also show an overall shift in level that depends on the microphone geometry. This latter result can be attributed to differences in the effective acoustic centers of the microphones. Using a Type M microphone and moving the calibration point farther from the lip ring of the artificial mouth substantially reduces the difference between the indicated and true free-field sound pressures. IEEE Standard 269–1971, “Method for Measuring Transmission Performance of Telephone Sets,” is being revised to implement these changes.
Free field
Noise-canceling microphone
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Time delay can't be avoided between reference audio signal and test signal in the audio quality evaluation named PEAQ,especially which is employed for monitoring online audio system.Calculating the time delay and making time alignment between reference audio signal and test signal is very important for audio quality evaluation in audio coding,audio system and audio communication.To deal with the problem mentioned above,the article presents a real-time and efficient time delay algorithm by calculating audio envelope,frequency spectrum cross correlation and histogram.When this algorithm is applied in the broadcast audio monitoring system,it shows the characteristic of real-time,steady and highly active.
Sound Quality
Audio signal flow
SIGNAL (programming language)
Spectral envelope
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Ambient underwater acoustics data are presented for one year at a potential tidal energy site in Admiralty Inlet, WA (USA) with maximum currents exceeding 3 m/s. The site, at a depth of approximately 60 meters, is located near shipping lanes, a local ferry route, and a transit area for many cetacean species. A key finding is that the statistical distribution of total sound pressure levels are dependent on tidal currents at the site. Pseudosound, cobbles shifting on the sea bed, and vibrations induced by forces on the equipment are possible explanations. Non-propagating turbulent pressure fluctuations, termed pseudosound, can mask ambient noise, especially in highly energetic environments suitable for tidal energy development. A statistical method identifies periods during which changes in the mean and standard deviation of the one-third octave band sound pressure levels are statistically significant and thus suggestive of pseudosound contamination. For each deployment, recordings with depth averaged tidal currents greater than 1 m/s are found to be contaminated, and only recordings with currents below this threshold are used in the subsequent ambient noise analysis. Mean total sound pressure levels (0.156 - 30 kHz) over all recordings are 117 dB re 1μPa. Total sound pressure levels exceed 100 dB re 1μPa 99% of the time and exceed 135 dB re 1μPa 4% of the time. Commercial shipping and ferry traffic are found to be the most significant contributors to ambient noise levels at the site, with secondary contributions from rain, wind, and marine mammal vocalizations. Post-processed data from an AIS (Automatic Identification System) receiver is used to determine the location of ships during each recording. Referencing 368 individual recordings with the distance between the ferry and the site obtained from AIS data, the source level of the ferry is estimated to be 179 ± 4 dB re 1μPa at 1m with a logarithmic spreading loss coefficient of 18.
Ambient noise level
Sound energy
Ambient pressure
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