Microphone Output Signal Research Articles

A Non-Linear Lumped Model for the Electro-Mechanical Coupling in Capacitive MEMS Microphones

This work provides an analytical non-linear model for the capacitive transduction in MEMS transducers with perforated counter-electrodes, especially applicable to capacitive MEMS microphones. Starting from an electrostatic description of a perforated unit cell of the transducer, analytical formulations of the variable capacitance and electrostatic forces are derived, accounting for the deflection profile of a clamped circular plate. A lumped implementation into conventional circuit simulations tools is enabled via behavioral modeling based on hardware description languages, such as Verilog-A. Therefore, the analytical model is approximated via Taylor series expansions, allowing for a stable and non-linear behavioral implementation. The resulting model finally enables both a small- and large-signal analysis of capacitive MEMS microphones, precisely accounting for non-linearities in the capacitive transduction. This allows to simulate the harmonic distortion of the microphone's output signal and to account for electrostatic spring-softening in simulations of its bias voltage dependent sensitivity.

Inaudible Attack on Smart Speakers With Intentional Electromagnetic Interference

This article demonstrates an inaudible attack on smart speakers using electromagnetic interference (EMI). The EMI induces voltages on the order of a few millivolts on conductors, which are then converted into baseband signals by exploiting the inherent nonlinearity of microphones. The EMI signal is specially preprocessed to minimize the useless harmonics generation at the microphone output signals, which significantly improves the recognition rate as well as nullify the previous countermeasures based on the harmonics detection. The sensitive carrier frequency found by our proposed method can improve the attack distance as well. A measurement-based methodology is applied to locate the sensitive regions for noise coupling without knowing the layout of the printed circuit board (PCB), and the transfer function is also obtained to insure the main coupling location. Our experiments show that in open space, intentional EMI under 2.5 W can inject commands at distances up to 2.5 m on smart speakers.

Bone-Conduction Hearing Aids

Bone-Conduction Hearing Aids

Low Distortion Analog Front-End for Digital Electret Microphone

This paper presents a low distortion analog front-end (AFE) circuit to process electret microphone output signal. A source follower is employed for the input buffer to interface electret microphone directly to the IC with level shifting. A single-ended to differential converter with output common-mode control is presented to compensate the common-mode variation resulted from gate to source voltage variation in the source follower. A replica stage is adopted to control the output bias voltage of the single-ended to differential converter. The prototype AFE circuit fabricated in a 0.35μm CMOS technology achieves 68.2dB peak SNDR and 79.9dB SFDR over an audio signal bandwidth of 20kHz with 2.5V supply while consuming 1.05mW.

HEARING INSTRUMENT WITH IMPROVED INITIALISATION OF PARAMETERS OF DIGITAL FEEDBACK SUPPRESSION CIRCUITRY

A method of modelling a feedback path from a receiver to a microphone in a hearing instrument, includes applying an electronic probe signal with an increasing level as a function of time to the receiver for conversion of the electronic probe signal into an acoustic probe signal output by the receiver, monitoring values of a first quality parameter calculated based at least in part on recorded microphone output signal, and after the first quality parameter reaches a predetermined first threshold value, determining at least one parameter of the feedback path based at least in part on the recorded microphone output signal values.

Wind noise at microphones within and across hearing aids at wind speeds below and above microphone saturation

The variation of wind noise at hearing-aid microphones with wind speed, wind azimuth, and hearing-aid style was investigated. Comparisons were made across behind-the-ear (BTE) and completely-in-canal (CIC) devices, and between microphones within BTE devices. One CIC device and two BTE devices were placed on a Knowles Electronics Manikin for Acoustic Research. The smaller BTE device had vented plastic windshields around its microphone ports while the larger BTE device had none. The microphone output signals were digitally recorded in wind generated at 0, 3, 6, and 12 m/s at 8 wind azimuths. The microphone output signals were saturated at 12 m/s with wind-noise levels of up to 116 dB SPL at the microphone output. Wind-noise levels differed by up to 12 dB between microphones within the same BTE device, and across BTE devices by up to 6 or 8 dB for front or rear microphones, respectively. On average, wind-noise levels were lowest with the CIC device and highest at the rear microphone of the smaller BTE device. Engineering and clinical implications are discussed.

Active Vibration Attenuation For Implantable Microphone

The invention is directed to an implanted microphone having reduced sensitivity to vibration. In this regard, the microphone differentiates between the desirable and undesirable vibration by utilizing at least one motion sensor to produce a motion signal when an implanted microphone is in motion. This motion signal is used to yield a microphone output signal that is less vibration sensitive. In a first arrangement, the motion signal may be processed with an output of the implantable microphone transducer to provide an audio signal that is less vibration-sensitive than the microphone output alone. Specifically, the motion signal may be scaled to match the motion component of the microphone output such that upon removal of the motion signal from the microphone output, the remaining signal is an acoustic signal.

Dual feedback control system for implantable hearing instrument

The invention is directed to an implanted microphone having reduced sensitivity to vibration. In this regard, the microphone differentiates between the desirable and undesirable vibration by utilizing at least one motion sensor to produce a motion signal when an implanted microphone is in motion. This motion signal is used to yield a microphone output signal that is less vibration sensitive. In a first arrangement, the motion signal may be processed with an output of the implantable microphone transducer to provide an audio signal that is less vibration-sensitive than the microphone output alone. Specifically, the motion signal may be scaled to match the motion component of the microphone output such that upon removal of the motion signal from the microphone output, the remaining signal is an acoustic signal.

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.

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.

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.

Acoustic rendering of a virtual environment based on virtual microphone control and binaural room scanning

Binaural room scanning (BRS) is a convolution technique that utilizes a set of spatially indexed impulse responses that are selected in response to listener head movements during virtual acoustic rendering, such that the direction of sonic elements is automatically updated according to the angle determined by a head tracker. Since the room impulse responses have to be measured for only a few loudspeaker positions, a very successful application of BRS is the simulation of a control room. In order to create a flexible headphone‐based virtual environment, it is proposed to simulate the input signals for the BRS system using virtual microphone control (ViMiC). ViMiC renders a virtual environment by simulating the output signals of virtual microphones that can be used to address a surround loudspeaker system. The advantages of this approach are twofold. First, the measured impulse responses of the BRS system ensure high spatial density in overall reflection patterns, avoiding the typical gaps in between the early reflections that occur when using solely mirror images or ray tracing. Second, the resulting imaging is closer to common audio engineering practice, where a pleasant but plausible environment is favored over a more purely realistic rendering.

An ultrasonic obstacle detector based on phase beamforming principles

A prototype of an ultrasonic obstacle detector developed as a part of a navigation system for blind and visually impaired people is presented. The detector, which employs a single ultrasound source and an array of microphones, determines the distance to the obstacle by measuring time between sending the pulse and receiving the reflected signal. Using the phase beamforming technique, borrowed from hydroacoustics, to process the output signals of microphones, it determines also the direction from which the reflected signal is received, thereby locating the obstacle. The obstacle detector was subjected to a series of tests in order to verify its design and to assess its ability to detect a broad range of obstacles. Presented test results show that most obstacles can be detected and recognized, and that the tested obstacle detector provides complete coverage of the safety zone in front of the user.

Directional microphone assembly

The present invention relates to a directional microphone assembly comprising a front and a rear microphone for compact communication devices, such as hearing aids, mobile phones etc. The present invention integrates most or all electronic processing circuits required to form a directional microphone output signal using a front and a rear microphone signal. Accordingly, the directional microphone assembly according to the present invention occupies less volume compared to traditional microphone assemblies.

Hearing aid with a directional microphone system as well as method for the operation thereof

In a hearing aid with a directional microphone system having a signal processing unit, an earphone and a number of microphones, the output signals of the respective microphones can be interconnected with one another with different weightings for generating an individual directional microphones characteristic by using one or more delay elements and the signal processing unit. The invention is also directed to a method for the operation of such a hearing aid.

Audio headset communication apparatus and method

An audio communications system enables a person using an stereophonic audio headset to selectively communicate simultaneously with two or more persons. The system includes a headset having a first earphone for reproducing a first voice communication in response to a received first audio communication signal and a second earphone for reproducing a second voice communication in response to a received second audio communication signal. Control means located for use by a person wearing the headset selectively enable the first voice communication to be reproduced by the first earphone and/or the second voice communication to be reproduced by the second earphone. Attention signals associated with the respective earphones are received and processed by the control means. A microphone output signal produced by a microphone on the headset can be selectively provided to a plurality of output terminals and an attention signal can be selectively provided with the microphone output signal to selected output terminals. A distinctive audio pattern is produced in one of the earphones to help a person wearing the audio headset to distinguish between the reproduced voices of two different persons that are respectively reproduced during the same interval by the different earphones.

A microphone array for hearing aids

A directional acoustic receiving system is constructed in the form of a necklace including an array of two or more microphones mounted on a housing supported on the chest of a user by a conducting loop encircling the user’s neck. Signal processing electronics contained in the same housing receives and combines the microphone signals in such a manner as to provide an amplified output signal which emphasizes sounds of interest arriving in a direction forward of the user. The amplified output signal drives the supporting conducting loop to produce a representative magnetic field. An electroacoustic transducer including a magnetic field pick-up coil for receiving the magnetic field is mounted in on the user’s ear and generates an acoustic signal representative of the sound of interest. The microphone output signals are weighted (scaled) and combined to achieve desired spatial directivity responses. The weighting coefficients are determined by an optimization process. By bandpass filtering the weighted microphone signals with a set of filters covering the audio frequency range and summing the filtered signals, a receiving microphone array with a small aperture size is caused to have a directivity pattern that is essentially uniform over frequency in two or three dimensions. This method enables the design of highly directive hearing instruments, which are comfortable, inconspicuous, and convenient to use. The invention provides the user with a dramatic improvement in speech perception over existing hearing aid designs, and particularly in the presence of background noise and reverberation.

A microphone array for hearing aids

A directional acoustic receiving system is a form of a necklace including an array of two or more microphones mounted on a housing supported on the chest of a user by a conducting loop encircling the user's neck. Signal processing electronics contained in the same housing receive and combine the microphone signals in such a manner as to provide an amplified output signal which emphasizes sounds of interest arriving in a direction forward of the user. The amplified output signal drives the supporting conducting loop to produce a representative magnetic field. An electroacoustic transducer including a magnetic field pick up coil for receiving the magnetic field is mounted in or on the user's ear and generates an acoustic signal representative of the sounds of interest. The microphone output signals are weighted (scaled) and combined to achieve desired spatial directivity responses. The weighting coefficients are determined by an optimization process. By bandpass filtering the weighted microphone signals, with a set of filters covering the audio frequency range, and summing the filtered signals, a receiving microphone array with a small aperture size is caused to have a directivity pattern that is essentially uniform over frequency in two or three dimensions. This method enables the design of highly-directive-hearing instruments which are comfortable, inconspicuous, and convenient to use. The array provides the user with a dramatic improvement in speech perception over existing hearing aid designs, particularly in the presence of background noise, reverberation, and feedback.

DETECTION OF PISTACHIO NUTS WITH CLOSED SHELLS USING IMPACT ACOUSTICS

An acoustical sorting system was developed to separate pistachio nuts with closed shells from those with open shells. The system includes a microphone, digital signal processing hardware, material handling equipment, and an air reject mechanism. It was found that upon impact with a steel plate, nuts with closed shells emit sound with higher signal magnitudes for the first 0.33 ms than do nuts with open shells. After this interval, nuts with closed shells emit sounds with lower signal magnitudes than those with open shells. Linear discriminant analysis was used to classify nuts using three features extracted from the microphone signal during the first 1.4 ms after impact. One of the discriminant features is the integrated absolute value of microphone output signal during the first 0.11 ms after impact. The other two features are the number of data points in the digitized microphone signal, between 0.6 and 1.4 ms after impact, that have a slope and signal magnitude below preset threshold levels. The classification accuracy of this system is approximately 97%. Throughput rate is approximately 40 nuts/s. Cost is about $7,000 to $10,000 per channel. This cost is much lower than that of color sorters used to remove other pistachio defects while throughput is comparable. Currently, closed–shell pistachio nuts are removed by mechanical devices. These devices have a lower classification accuracy (95%) and damage kernels in open–shell pistachios by “pricking” them with a needle. The needle hole can give the appearance of an insect tunnel and cause rejection by the consumer. The newly developed system does not cause such damage. Increased sorting accuracy of the acoustic sorter, coupled with low cost, enables a payback period of less than one year.

Internal noise in miniature electret microphones

Modern miniature electret microphones normally include an internal amplifier to buffer the very high electrical impedance of the electret motor. This amplifier, usually an FET source follower circuit, generates a part of the noise in the microphone output signal. The electrical noise originates in the semiconductor and in the resistive components of the electrical circuit. Recent developments have reduced the electrical noise considerably so that it is no longer the dominant source. The other noise sources are the resistive components of the mechanical and acoustical impedances within the microphone. This paper describes recent work and special measurement methods that have helped to identify and reduce the acoustical noise components.