Microphone Output Research Articles

マイクロフォンによる配管外部からのキャビテーション検知(機械力学,計測,自動制御)

Cavitation induced vibration and erosion of pipes are the potential damaging factors in a piping system. To prevent such trouble, it is necessary to develop the detection method of cavitation phenomena. Especially, in power plants, it is preferred to detect them from the outside of piping during operation. In this paper, detection of cavitation phenomena was experimentally investigated using microphones placed outside of piping at upstream and downstream of an orifice. Following results were obtained: (1) According to the progress of cavitation phase, output voltage of the microphone varied, and the amplitude and number of the pulse-shaped signals increased. However, it migh be difficult to distinguish them from surrounding noises in an operating plant. (2) Microphone outputs were confirmed to be radiasion sounds caused by vibration on surface of the piping based on measurements of time difference between accelerometer output and microphone output. (3) The results of 1/3 octave band analysis revealed, noises by cavitation increased in the region of high frequency accoding to the progress of cavitation. In transitional phase, high frequency noises at the downstream from the orifice were larger than upstream. (4) The RMS (root mean square) values ratio of microphone output at upstream and downstream of the orifice varied according to the progress of cavitation, and increased by applying a high band pass filter. Therefore, comparison of RMS values at upstream and downstream of the orifice might make it possible to detect cavitation phenomena in a piping system of operating plants.

DETECTING HOLLOW HEART OF POTATO TUBERS USING IMPACT SOUND

An acoustical sorting system was developed to detect hollow heart (cavity) of potato tubers (Spun-ta variety). The system includes a microphone and digital signal processing hardware. It was found that upon impact with a steel plate, potato tubers with no hollow heart emit sound with higher signal magnitudes than with hollow one. Linear discrimi-nate analysis was used to classify potato tubers using three features extracted from the micro-phone signal. One of the discriminate features is the integrated absolute value of microphone output signal. The other two features are the number of data points in the digitized microphone signal after impact that have a slope and signal magnitude below preset threshold levels. The classification accuracy of this system is approximately 98%. The internal quality of potato tubers can be detected by acoustic impact method. The resonant frequency of potato tubers after dropping was found from acquired signal. The hollow heart existence of po-tato tubers can be evaluated by the resonant fre-quency of vibration signal of dropping tubers. This technology of acoustics to evaluate the hollow heart existence is more reliable and accurate spe-cially for export.

Speech separation based on law of causality

This paper proposes a microphone array system separating speech signals, based on a different principle from independent component analysis (ICA). This system applies linear prediction error filters to microphone outputs, and using the prediction errors, adjusts the coefficients of adaptive filters. In this case, only the prediction errors satisfying the low of causality become available for the adjustment; consequently, this system can steer a null toward a direction satisfying it. The permutation problem discussed in ICA can be thereby avoided. This system also can compensate the separated speech signals by using adaptive filters, and finally, can provide high quality speech signals. This paper moreover verifies the performance of the proposed system by using speech signal data measured in an ordinary room. This result shows that the proposed system works well even in reverberation environment.

An acoustic impact method to detect hollow heart of potato tubers

An acoustic sorting system was developed to detect hollow heart in potato tubers (Spunta variety). The system includes a microphone, digital signal processing hardware and material handling equipment. Upon impact with a steel plate, it was found that solid potato tubers emitted higher magnitude sounds than hollow tubers. Linear discriminate analysis was used to classify potato tubers using three features extracted from the microphone signal. One of the discriminate features was the integrated absolute value of microphone output signal. The other two features were the number of data points in the digitised microphone signal following impact that have slopes and magnitudes below preset threshold levels. The classification accuracy of this system was approximately 98%. Thus, it was concluded that the internal quality of potato tubers can be detected by the acoustic impact method.

Noninvasive method for determining blood pressure and contours of arterial and volume pulses

A noninvasive method for monitoring blood pressure, based on the principles established by Riva-Rocci and Korotkoff (K), is described; it furnishes, after a single compression-deflation cycle of the arm-encircling cuff, values of sys-tolic and diastolic blood pressures as well as the contours of the brachial arterial pulse and the corresponding volume pulse. K-sounds are detected by a single microphone situated in the cubital fossa, and the time-varying cuff pressure P(t) is read by a piezoresistive pressure sensor. The behavior of P(t) during deflation is resolved into two parts, P(t)=p(t)+b(t); p is a train of posi-tive going pulses (arising from arterial pulsa-tions), whereas b is a slowly changing baseline. Noise pulses in the microphone output are re-jected by using the observation that the first few K-sounds are emitted when p is close to a maxi-mum, and the last few when dp/dt is close to a maximum. The performance of the instrument is illustrated by showing how it copes with ambi-ent noise and involuntary manual perturbations of P, and by presenting contours of various pulses.

Acoustic Multitarget Tracking Using Direction-of-Arrival Batches

In this paper, we propose a particle filter acoustic direction-of-arrival (DOA) tracker to track multiple maneuvering targets using a state space approach. The particle filter determines its state vector using a batch of DOA estimates. The filter likelihood treats the observations as an image, using template models derived from the state update equation, and also incorporates the possibility of missing data as well as spurious DOA observations. Multiple targets are handled using a partitioned state-vector approach. The particle filter solution is compared with three other methods: the extended Kalman filter, Laplacian filter, and another particle filter that uses the acoustic microphone outputs directly. In addition, we demonstrate an autonomous system for multiple target DOA tracking with automatic target initialization and deletion. The initialization system uses a track-before-detect approach and employs matching pursuit to initialize multiple targets. Computer simulations are presented to compare the performance of the algorithms

Effect of Power Line Interference on Microphone Calibration Measurements Made at or Near Harmonics of the Power Line Frequency.

The electrical measurements required during the primary calibrations of laboratory standard microphones by the reciprocity method can be influenced by power line interference. Because of this influence, the protocols of international inter-laboratory key comparisons of microphone calibrations usually have not included measurements at power line frequencies. Such interference has been observed in microphone output voltage measurements made with a microphone pressure reciprocity calibration system under development at NIST. This system was configured for a particular type of standard microphone in such a way that measurements of relatively small signal levels, which are more susceptible to the effect of power line interference, were required. This effect was investigated by acquiring microphone output voltage measurement data with the power line frequency adjusted to move the frequency of the interference relative to the center frequency of the measurement system passband. These data showed that the effect of power line interference for this system configuration can be more than one percent at test frequencies harmonically related to the power line frequency. These data also showed that adjusting the power line frequency to separate the interference and test frequencies by as little as 1.0 Hz can reduce the effect of the interference by at least an order of magnitude. Adjustment of the power line frequency could enable accurate measurements at test frequencies that otherwise might be avoided.

Joint noise reduction and acoustic echo cancellation using the transfer-function generalized sidelobe canceller

Man machine interaction requires an acoustic interface for providing full duplex hands-free communication. The transfer-function generalized sidelobe canceller (TF-GSC) is an adaptive beamformer suitable for enhancing a speech signal received by an array of microphones in a noisy and reverberant environment. When an echo signal is also present in the microphone output signals, cascade schemes of acoustic echo cancellation and TF-GSC can be employed for suppressing both interferences. However, the performances obtainable by cascade schemes are generally insufficient. An acoustic echo canceller (AEC) that precedes the adaptive beamformer suffers from the noise component at its input. Acoustic echo cancellation following the adaptive beamformer lacks robustness due to time variations in the echo path affecting beamformer adaptation. In this paper, we introduce an echo transfer-function generalized sidelobe canceller (ETF-GSC), which combines the TF-GSC with an acoustic echo canceller. The proposed scheme consists of a primary TF-GSC for dealing with the noise interferences, and a secondary modified TF-GSC for dealing with the echo cancellation. The secondary TF-GSC includes an echo canceller embedded within a replica of the primary TF-GSC components. We show that using this structure, the problems encountered in the cascade schemes can be appropriately avoided. Experimental results demonstrate improved performance of the ETF-GSC compared to cascade schemes in noisy and reverberant environments.

Apparatus and method for matching the response of microphones in magnitude and phase

An apparatus is provided for matching the response of a pair of microphones. The two microphones provide a first and second output, respectively, in response to an audible input. The microphone outputs are subtract from each other to produce a gain control output for operably controlling the gain of the first microphone output, resulting in a gain compensated microphone output. A phase adjustment circuit also is provided responsive to the gain compensated microphone output and a rolloff control output for producing a matching output. The rolloff control output is generated by a phase difference subtractor circuit responsive to both the matching output and the second microphone output. Moreover, the output of at least one of the microphones has a resonance frequency that is shifted to a desired preselected frequency.

Aided ear bud

A hearing aid device is configured to operate as an ear bud device for a wireless phone. The hearing aid device functions in a manner similar to a conventional hearing aid device in a first operational state. The operational state of the hearing aid device is changed to a second operational state, in some embodiments, by a verified input from a wireless phone. Upon the change in operational state, the microphone input from the aided ear bud is output to the wireless phone while the incoming transmission from the wireless phone is combined with the microphone input and output to the user. The hearing aid device can be modified to receive analog or digital inputs from other outside sources. Exemplary methods for operating a hearing aid device is disclosed. An exemplary mobile communications device for interfacing with the hearing aid device also is disclosed.

Frequency domain simultaneous equations method for active noise control systems

This study proposes a new method for feedforward‐type active noise control systems. This method, named the frequency domain simultaneous equations method, is based on a different principle from the filtered‐x algorithm requiring a filter modeled on a secondary path from a loudspeaker to an error microphone. Instead of the filter, this method uses an auxiliary filter identifying the overall path consisting of a primary path, a noise control filter, and the secondary path. As seen from the configuration of the overall path, the auxiliary filter can provide two independent equations when two different coefficient vectors are given to the noise control filter. In practical use, the two different coefficient vectors are provided by their own estimation errors. By solving the independent equations, the proposed method can estimate the coefficient vector of the noise control filter minimizing the output of the error microphone. This study first presents computer simulation results demonstrating that the proposed method can automatically recover the noise reduction that is in effect degraded by path changes. Finally, this study applies the proposed method to an experimental system and, by using the system, verifies that the proposed method is functional in practical systems.

MICRO ACOUSTIC DEVICES USING PIEZOELECTRIC FILMS

ABSTRACT A novel PZT based micro acoustic device with the diaphragms clamped on all four edges has been studied. It can be used both as microphone and microspeaker. Compared with other piezoelectric micro acoustic devices, PZT based device has higher sensitivity for microphone and larger output acoustic pressure for microspeaker. The microfabrication process flow of this device is simple, and the transducer has excellent performance, miniature size and high reliability. The micro acoustic devices could be widely used in various practical audio frequencies and ultrasonic systems.

Partial differential equation (PDE)-based method of sound source localization

We describe the problem of direction and distance estimate of sound sources in free 3-D space. We first derive a partial differential equation (PDE), ∇f(r,t)={(1/R)f(r,t)+(1/c)ḟ(r,t)}n(R≡|r|,n≡-r/R), what we call the sound source constraint (SSC). We show that the general solution of the SSC is a diverging spherical wave from a point sounce with arbitrary temporal waveform. The SSC enables the observer to determine the source location (distance R and direction n) from local measurements of the wave field. As the measurements of the wave field, we consider two methods: (1) the weighted temporal integral of arrayed microphone outputs in a finite duration and (2) the weighted spatial integral of pressure distribution in a rectangular or circular area. In both cases, we obtain exact formulas for localizing a single source from a single measurement and multiple sources from the combination of differently weighted measurements.

Non-contact rotating beam crack size estimation from vibro-acoustic signals

This paper describes a new and practical method to estimate the size of a crack on a rotating beam in a laboratory setting. The paper consists of selecting and validating a sensor and a measurement variable, devising a signal processing method for crack size estimation and carrying out experimental validations. The study employed a microphone to measure the pressure wave excited by the vibration of the rotating beam, validated the microphone signal, utilized chirp z-transform to extract the first and second vibration mode frequencies and established a diagnostic neural network to map the frequencies to crack size. Four fatigue tests were conducted to initiate and then propagate a crack. Microphone outputs and crack sizes were periodically recorded during the four tests. Data of test no. 1 were used to calibrate the neural net and data of the other three tests were used for testing. The experimental results show that the proposed approach can provide reasonably good estimates of the crack size using the indirectly excited acoustic signal.

Contribution of cross talk to the uncertainty of microphone calibrations by the electrostatic actuator method

Cross talk in electrostatic actuator calibrations is defined as the response of a test microphone to the actuator excitation voltage when the microphone membrane is frozen, i.e., when the actuator polarization voltage is zero—an unavoidable nuisance effect that contributes to the calibration uncertainty. Two models for the cross talk are considered. In the first, the actuator excitation voltage is coupled to the microphone output through the stray capacitance between the actuator electrode and the microphone backplate; in the second, it is coupled through the ground resistance, which provides a common element between the input and output loops. In the stray capacitance model, at frequencies below the membrane resonance, the cross talk is independent of frequency, but in the ground resistance model it varies inversely with frequency. The frequency dependence of the cross talk thus serves as a discriminator between the two models. Measurements on condenser microphones of various sizes yield results that favor the stray capacitance model. The ratio of cross-talk signal to actuator pressure signal (at 94 dB re: 20 mPa) is found experimentally to be −74 dB for a 1-in. condenser microphone and nearly the same for 1/2-in. and 1/4-in. microphones.

Cochlear Implant Speech Processor Placement and Compression Effects on Sound Sensitivity and Interaural Level Difference

The purpose of this investigation was to determine the impact of commonly recommended cochlear implant (CI) speech processor placements on microphone output both with and without single channel front-end compression. The impact of this compression use on interaural level difference (ILD) magnitude was also evaluated for the ear-level position. Finally, pilot localization data collected with and without single channel front-end compression was collected on seven bilateral cochlear implant recipients. The results revealed that differences in signal audibility due to clinical placement of CI speech processors in ear, shoulder, and collar positions can at least partially be offset through the use of front-end compression. These data also revealed that compression impacted ILD cues. Preliminary data indicated that some bilaterally implanted subjects were able to take advantage of the enhanced ILD cues when compression was turned off, while other bilaterally implanted subjects did not localize better in the compression-off condition.

Verification of simultaneous equations method by an experimental active noise control system

In this study, we verify the performance of the simultaneous equations method using an experimental active noise control system. The simultaneous equations method is based on a priciple different from the filtered-x algorithm requiring a filter modeled on a secondary path from a loudspeaker to an error microphone. Instead of the filter, called the secondary path filter, this method uses an auxiliary filter identifying the overall path consisting of a primary path, a noise control filter and the secondary path. As inferred from the configuration of the overall path, the auxiliary filter can provide two independent equations when two different coefficient vectors are given to the noise control filter. The method thereby estimates the coefficient vector of the noise control filter minimizing the output of the error microphone. In this paper, we propose the application of a frequency domain adaptive algorithm to the identification of the overall path. An improvement in the noise reduction speed is thereby expected. In this paper, we also present computer simulation results demonstrating that the simultaneous equations method can automatically recover the noise reduction effect degraded by path changes, and finally, using an experimental system, we indicate that the method successfully works in practical systems.

Processing of reverberant speech for time-delay estimation

In this paper, we present a method of extracting the time-delay between speech signals collected at two microphone locations. Time-delay estimation from microphone outputs is the first step for many sound localization algorithms, and also for enhancement of speech. For time-delay estimation, speech signals are normally processed using short-time spectral information (either magnitude or phase or both). The spectral features are affected by degradations in speech caused by noise and reverberation. Features corresponding to the excitation source of the speech production mechanism are robust to such degradations. We show that these source features can be extracted reliably from the speech signal. The time-delay estimate can be obtained using the features extracted even from short segments (50-100 ms) of speech from a pair of microphones. The proposed method for time-delay estimation is found to perform better than the generalized cross-correlation (GCC) approach. A method for enhancement of speech is also proposed using the knowledge of the time-delay and the information of the excitation source.

Separation of Sound Sources Propagated in the Same Direction

This paper describes a method for separating a target sound from other noise arriving in a single direction when the target cannot, therefore, be separated by directivity control. Microphones are arranged in a line toward the sources to form null sensitivity points at given distances from the microphones. The null points exclude non-target sound sources on the basis of weighting coefficients for microphone outputs determined by blind source separation. The separation problem is thereby simplified to instantaneous separation by adjustment of the time-delays for microphone outputs. The system uses a direct (i.e. non-iterative) algorithm for blind separation based on second-order statistics, assuming that all sources are non-stationary signals. Simulations show that the 2-microphone system can separate a target sound with separability of more than 40 dB for the 2-source problem, and 25 dB for the 3-source problem when the other sources are adjacent.

Laser Doppler velocimetry based on the photoacoustic effect in a CO2 laser

We report a simple laser Doppler velocimeter in which the photoacoustic effect was used to measure the rotation wheel speed. A Doppler signal, caused by mixing a returning wave with an originally existing wave inside the CO2 laser cavity, was detected using a microphone in the laser tube. Frequency of the microphone output was in proportion to the rotation speed of a wheel and is dependent on the cosine of the angle between the direction of the laser beam and tangent of wheel velocity. A Doppler-shifted frequency as high as 34kHz was detected using this method. A frequency response of a few megahertz is expected from the laser Doppler velocimeter based on the photoacoustic effect in a CO2 laser by using a wider bandwidth microphone.