Showing papers on "Microphone published in 2015"

Ultrathin, Rollable, Paper-Based Triboelectric Nanogenerator for Acoustic Energy Harvesting and Self-Powered Sound Recording

Abstract: A 125 μm thickness, rollable, paper-based triboelectric nanogenerator (TENG) has been developed for harvesting sound wave energy, which is capable of delivering a maximum power density of 121 mW/m(2) and 968 W/m(3) under a sound pressure of 117 dBSPL. The TENG is designed in the contact-separation mode using membranes that have rationally designed holes at one side. The TENG can be implemented onto a commercial cell phone for acoustic energy harvesting from human talking; the electricity generated can be used to charge a capacitor at a rate of 0.144 V/s. Additionally, owing to the superior advantages of a broad working bandwidth, thin structure, and flexibility, a self-powered microphone for sound recording with rolled structure is demonstrated for all-sound recording without an angular dependence. The concept and design presented in this work can be extensively applied to a variety of other circumstances for either energy-harvesting or sensing purposes, for example, wearable and flexible electronics, military surveillance, jet engine noise reduction, low-cost implantable human ear, and wireless technology applications.

Phase Processing for Single-Channel Speech Enhancement: History and recent advances

Abstract: With the advancement of technology, both assisted listening devices and speech communication devices are becoming more portable and also more frequently used. As a consequence, users of devices such as hearing aids, cochlear implants, and mobile telephones, expect their devices to work robustly anywhere and at any time. This holds in particular for challenging noisy environments like a cafeteria, a restaurant, a subway, a factory, or in traffic. One way to making assisted listening devices robust to noise is to apply speech enhancement algorithms. To improve the corrupted speech, spatial diversity can be exploited by a constructive combination of microphone signals (so-called beamforming), and by exploiting the different spectro?temporal properties of speech and noise. Here, we focus on single-channel speech enhancement algorithms which rely on spectrotemporal properties. On the one hand, these algorithms can be employed when the miniaturization of devices only allows for using a single microphone. On the other hand, when multiple microphones are available, single-channel algorithms can be employed as a postprocessor at the output of a beamformer. To exploit the short-term stationary properties of natural sounds, many of these approaches process the signal in a time-frequency representation, most frequently the short-time discrete Fourier transform (STFT) domain. In this domain, the coefficients of the signal are complex-valued, and can therefore be represented by their absolute value (referred to in the literature both as STFT magnitude and STFT amplitude) and their phase. While the modeling and processing of the STFT magnitude has been the center of interest in the past three decades, phase has been largely ignored.

Graphene electrostatic microphone and ultrasonic radio

Abstract: We present a graphene-based wideband microphone and a related ultrasonic radio that can be used for wireless communication. It is shown that graphene-based acoustic transmitters and receivers have a wide bandwidth, from the audible region (20∼20 kHz) to the ultrasonic region (20 kHz to at least 0.5 MHz). Using the graphene-based components, we demonstrate efficient high-fidelity information transmission using an ultrasonic band centered at 0.3 MHz. The graphene-based microphone is also shown to be capable of directly receiving ultrasound signals generated by bats in the field, and the ultrasonic radio, coupled to electromagnetic (EM) radio, is shown to function as a high-accuracy rangefinder. The ultrasonic radio could serve as a useful addition to wireless communication technology where the propagation of EM waves is difficult.

Chirp signal-based aerial acoustic communication for smart devices

Abstract: Smart devices such as smartphones and tablet/wearable PCs are equipped with voice user interface, ie, speaker and microphone Accordingly, various aerial acoustic communication techniques have been introduced to utilize the voice user interface as a communication interface In this paper, we propose an aerial acoustic communication system using inaudible audio signal for low-rate communication in indoor environments By adopting chirp signal, which is widely used for radar applications due to its capability of resolving multi-path propagation, the proposed acoustic modem supports long-range communication independent of device characteristics over severely frequency-selective acoustic channel We also design a backend server architecture to compensate for the low data rate of chirp signal-based acoustic modem Via extensive experiments, we evaluate various characteristics of the proposed modem including multi-path resolution and multiple chirp signal detection We also verify that the proposed chirp signal can deliver data at 16 bps in typical indoor environments, where its maximum transmission range is drastically extended up to 25 m compared to the few meters of the previous research

Cameras and networked security systems and methods

Abstract: A system includes a remote server and a device having a wireless transceiver, microphone, and processing circuit. The processing circuit is configured to monitor the microphone for a siren signature. The processing circuit is configured to use the wireless transceiver to send information to the remote server in response to a detection of the siren signature. The remote server causes wireless cameras located near the device to capture a current image and to send the captured image to the remote server for further distribution.

Hands-free human machine interface responsive to a driver of a vehicle

Abstract: Human machine interfaces for use in a vehicle are provided. According to one implementation, a human machine interface (HMI) comprises a housing configured to be mounted on a vehicle. The HMI also includes a sensor, a microphone, and a speaker disposed in the housing. The sensor is configured to sense image input received from a driver of the vehicle. The microphone is configured to receive speech input received from the driver. The speaker is configured to provide audio output to the driver. The HMI further includes a processing device disposed within the housing and coupled with the sensor, microphone, and speaker. The processing device is configured to process the image input and speech input.

Performance analysis of the covariance subtraction method for relative transfer function estimation and comparison to the covariance whitening method

Abstract: Microphone array processing utilize spatial separation between the desired speaker and interference signal for speech enhancement. The transfer functions (TFs) relating the speaker component at a reference microphone with all other microphones, denoted as the relative TFs (RTFs), play an important role in beamforming design criteria such as minimum variance distortionless response (MVDR) and speech distortion weighted multichannel Wiener filter (SDW-MWF). Two common methods for estimating the RTF are surveyed here, namely, the covariance subtraction (CS) and the covariance whitening (CW) methods. We analyze the performance of the CS method theoretically and empirically validate the results of the analysis through extensive simulations. Furthermore, empirically comparing the methods performances in various scenarios evidently shows thats the CW method outperforms the CS method.

Relayed voice control of devices

Abstract: Voice input is received, via a microphone. The voice input is translated into a machine-understandable representation of the voice input. An operation performable by an audio/video device and corresponding to the machine-understandable representation of the voice input is selected. A signal code for the operation is broadcast, via a broadcast device, in a format recognizable by the audio/video device.

Mobile interface for loudspeaker optimization

Abstract: A system for providing an audio processing interface at a mobile device configured to detect an audio processor, present, via a user interface, a display screen to receive user input to initiate audio testing, iteratively present a series of testing screens, each including at least one instruction and test status, and present another instruction and test status in response to receiving and indicative of a successful sample at a previous microphone location.

Stretchable Loudspeaker using Liquid Metal Microchannel

Abstract: Considering the various applications of wearable and bio-implantable devices, it is desirable to realize stretchable acoustic devices for body-attached applications such as sensing biological signals, hearing aids, and notification of information via sound. In this study, we demonstrate the facile fabrication of a Stretchable Acoustic Device (SAD) using liquid metal coil of Galinstan where the SAD is operated by the electromagnetic interaction between the liquid metal coil and a Neodymium (Nd) magnet. To fabricate a liquid metal coil, Galinstan was injected into a micro-patterned elastomer channel. This fabricated SAD was operated simultaneously as a loudspeaker and a microphone. Measurements of the frequency response confirmed that the SAD was mechanically stable under both 50% uniaxial and 30% biaxial strains. Furthermore, 2000 repetitive applications of a 50% uniaxial strain did not induce any noticeable degradation of the sound pressure. Both voice and the beeping sound of an alarm clock were successfully recorded and played back through our SAD while it was attached to the wrist under repeated deformation. These results demonstrate the high potential of the fabricated SAD using Galinstan voice coil in various research fields including stretchable, wearable, and bio-implantable acoustic devices.

Augmented reality sound system

Abstract: An augmented reality sound systems is disclosed. An augmented reality sound system includes a at least one microphone for receiving ambient sound and a memory storing one or more augmented reality sound profiles and a respective set of processing parameters. The system further includes a processor coupled to the memory and configured to generate augmented ambient sound from the ambient sound by reproducing the ambient sound in conjunction with processed sound superimposed over the ambient sound as directed by one or more of the set of processing parameters retrieved from the memory based on a selected augmented reality sound profile.

Speaker location and microphone spacing invariant acoustic modeling from raw multichannel waveforms

Abstract: Multichannel ASR systems commonly use separate modules to perform speech enhancement and acoustic modeling. In this paper, we present an algorithm to do multichannel enhancement jointly with the acoustic model, using a raw waveform convolutional LSTM deep neural network (CLDNN). We will show that our proposed method offers ∼5% relative improvement in WER over a log-mel CLDNN trained on multiple channels. Analysis shows that the proposed network learns to be robust to varying angles of arrival for the target speaker, and performs as well as a model that is given oracle knowledge of the true location. Finally, we show that training such a network on inputs captured using multiple (linear) array configurations results in a model that is robust to a range of microphone spacings.

SymDetector: detecting sound-related respiratory symptoms using smartphones

Abstract: This paper proposes SymDetector, a smartphone based application to unobtrusively detect the sound-related respiratory symptoms occurred in a user's daily life, including sneeze, cough, sniffle and throat clearing SymDetector uses the built-in microphone on the smartphone to continuously monitor a user's acoustic data and uses multi-level processes to detect and classify the respiratory symptoms Several practical issues are considered in developing SymDetector, such as users' privacy concerns about their acoustic data, resource constraints of the smartphone and different contexts of the smartphone We have implemented SymDetector on Galaxy S3 and evaluated its performance in real experiments involving 16 users and 204 days The experimental results show that SymDetector can detect these four types of respiratory symptoms with high accuracy under various conditions

Multi-Microphone Speech Recognition Systems and Related Techniques

Abstract: A speech recognition system for resolving impaired utterances can have a speech recognition engine configured to receive a plurality of representations of an utterance and concurrently to determine a plurality of highest-likelihood transcription candidates corresponding to each respective representation of the utterance. The recognition system can also have a selector configured to determine a most-likely accurate transcription from among the transcription candidates. As but one example, the plurality of representations of the utterance can be acquired by a microphone array, and beamforming techniques can generate independent streams of the utterance across various look directions using output from the microphone array.

Voice controlled assistant with coaxial speaker and microphone arrangement

Abstract: A voice controlled assistant has a housing to hold one or more microphones, one or more speakers, and various computing components. The housing has an elongated cylindrical body extending along a center axis between a base end and a top end. The microphone(s) are mounted in the top end and the speaker(s) are mounted proximal to the base end. The microphone(s) and speaker(s) are coaxially aligned along the center axis. The speaker(s) are oriented to output sound directionally toward the base end and opposite to the microphone(s) in the top end. The sound may then be redirected in a radial outward direction from the center axis at the base end so that the sound is output symmetric to, and equidistance from, the microphone(s).

Drone detection and classification methods and apparatus

Abstract: A system, method, and apparatus for drone detection and classification are disclosed. An example method includes receiving a sound signal in a microphone and recording, via a sound card, a digital sound sample of the sound signal, the digital sound sample having a predetermined duration. The method also includes processing, via a processor, the digital sound sample into a feature frequency spectrum. The method further includes applying, via the processor, broad spectrum matching to compare the feature frequency spectrum to at least one drone sound signature stored in a database, the at least one drone sound signature corresponding to a flight characteristic of a drone model. The method moreover includes, conditioned on matching the feature frequency spectrum to one of the drone sound signatures, transmitting, via the processor, an alert.

Multi-microphone speech dereverberation and noise reduction using relative early transfer functions

Abstract: In speech communication systems, the microphone signals are degraded by reverberation and ambient noise. The reverberant speech can be separated into two components, namely, an early speech component that includes the direct path and some early reflections, and a late reverberant component that includes all the late reflections. In this paper, a novel algorithm to simultaneously suppress early reflections, late reverberation and ambient noise is presented. A multi-microphone minimum mean square error estimator is used to obtain a spatially filtered vaersion of the early speech component. The estimator constructed as a minimum variance distortionless response (MVDR) beam-former (BF) followed by a postfilter (PF). Three unique design features characterize the proposed method. First, the MVDR BF is implemented in a special structure, named the nonorthogonal generalized sidelobe canceller (NO-GSC). Compared with the more conventional orthogonal GSC structure, the new structure allows for a simpler implementation of the GSC blocks for various MVDR constraints. Second, In contrast to earlier works, RETFs are used in the MVDR criterion rather than either the entire RTFs or only the direct-path of the desired speech signal. An estimator of the RETFs is proposed as well. Third, the late reverberation and noise are processed by both the beamforming stage and the PF stage. Since the relative power of the noise and the late reverberation varies with the frame index, a computationally efficient method for the required matrix inversion is proposed to circumvent the cumbersome mathematical operation. The algorithm was evaluated and compared with two alternative multichannel algorithms and one single-channel algorithm using simulated data and data recorded in a room with a reverberation time of 0.5 s for various source-microphone array distances (1-4 m) and several signal-to-noise levels. The processed signals were tested using two commonly used objective measures, namely perceptual evaluation of speech quality and log-spectral distance. As an additional objective measure, the improvement in word accuracy percentage of an acoustic speech recognition system is also demonstrated.

Investigation of a Sweep Technique for Microphone Placement

Abstract: Investigation of a Sweep Technique for Microphone Placement C.P.F. Verster Department of Music, University of Stellenbosch, Private Bag X1, Matieland 7602, South Africa. Thesis: MPhil Music Technology November 2014 Microphone placement is a tedious, time consuming, trial and error based process. Recordists use microphone placement as a method for altering the recorded timbre of an instrument and it is a vital part of the recording process. This thesis investigated a microphone sweep technique and remote microphone positioning as methods for identifying and evaluating microphone placements. iii Stellenbosch University https://scholar.sun.ac.za

Playback device calibration

Abstract: Systems and methods are provided for device playback calibration. An example implementation involves a computing device receiving, via a microphone, detected audio content rendered by at least one playback device. The implementation also involves the computing device modulating the detected audio content with a modulation signal such that the modulation signal has a modulation frequency determined based on an input frequency range of a processing unit. The implementation also involves providing the modulated audio content to the processing unit; and determining, via the processing unit, an equalization setting for the at least one playback device.

Microphone array method for the characterization of rotating sound sources in axial fans

Abstract: Methods based on microphone array measurements provide a powerful tool for determining the location and magnitude of acoustic sources. For stationary sources, sophisticated algorithms working in the frequency domain can be applied. By using circularly arranged arrays and interpolating between microphone signals it is possible to treat rotating sources, as are present in fans, as being non-moving. Measurements conducted with a four-bladed fan and analyzed with the “virtual rotating array” method show that it is not only possible to identify the main noise contributors, but also to determine a full spectrum for any rotating component of interest.

A MEMS Low-Noise Sound Pressure Gradient Microphone With Capacitive Sensing

Abstract: A silicon microelectromechanical systems microphone is described that detects sound pressure gradients. The diaphragm consists of a stiffened plate that rotates around a central axis in response to sound pressure gradients. The motion of the diaphragm is converted into an electronic signal through the use of interdigitated comb fins that enable capacitive sensing. Measured results show that the microphone achieves a substantially lower low-frequency sound pressure-referred noise floor than can be achieved using existing dual miniature microphone systems. Measured directivity patterns are shown to be very close to what is expected for sound pressure gradient receivers over a broad range of frequencies.

Self-calibration of multi-microphone noise reduction system for hearing assistance devices using an auxiliary device

Abstract: The application relates to hearing assistance system (use and method) for calibrating a noise reduction system of a hearing assistance device. The system comprises a hearing assistance device, and an auxiliary device. The hearing assistance device comprises a multitude of input units, and a multichannel beamformer filtering unit configured to determine filter weights for a beamformed signal. The system further comprises a user interface for activating a calibration mode. The auxiliary device comprises an output transducer for converting an electric calibration signal to an acoustic calibration sound signal. The system is configured to estimate a look vector for a target signal originating from a target signal source located at a specific location relative to the user based on the acoustic calibration sound signal.

Multilayer graphene condenser microphone

Abstract: Vibrating membranes are the cornerstone of acoustic technology, forming the backbone of modern loudspeakers and microphones. Acoustic performance of a condenser microphone is derived mainly from the membrane's size, surface mass and achievable static tension. The widely studied and available nickel has been a dominant membrane material for professional microphones for several decades. In this paper we introduce multilayer graphene as a membrane material for condenser microphones. The graphene device outperforms a high end commercial nickel-based microphone over a significant part of the audio spectrum, with a larger than 10 dB enhancement of sensitivity. Our experimental results are supported with numerical simulations, which also show that a 300 layer thick graphene membrane under maximum tension would offer excellent extension of the frequency range, up to 1 MHz.

Modeling local active sound control with remote sensors in spatially random pressure fields

Abstract: A general formulation is presented for the optimum controller in an active system for local sound control in a spatially random primary field. The sound field in a control region is selectively attenuated using secondary sources, driven by reference sensors, all of which are potentially remote from this control region. It is shown that the optimal controller is formed of the combination of a least-squares estimation of the primary source signals from the reference signals, and a least-squares controller driven by the primary source signals themselves. The optimum controller is also calculated using the remote microphone technique, in both the frequency and the time domains. The sound field under control is assumed to be stationary and generated by an array of primary sources, whose source strengths are specified using a spectral density matrix. This can easily be used to synthesize a diffuse primary field, if the primary sources are uncorrelated and far from the control region, but can also generate primary fields dominated by contributions from a particular direction, for example, which is shown to significantly affect the shape of the resulting zone of quiet.

Use of an earpiece acoustic opening as a microphone port for beamforming applications

Abstract: A mobile multi-function device that includes a speaker, two or more microphones, and a beamformer processor is described. The beamformer processor uses the microphones to perform beamforming operations. One of the microphones shares a receiver acoustic opening with the speaker while the other microphone uses a separate acoustic opening. The receiver acoustic opening may be an earpiece opening that is held to the ear of a user while conducting a phone call with the device and provides acoustic input and output paths for the microphone and the speaker, respectively.

Multilayer graphene condenser microphone

Abstract: Vibrating membranes are the cornerstone of acoustic technology, forming the backbone of modern loudspeakers and microphones. Acoustic performance of condenser microphone is derived mainly from the membrane's size and achievable static tension. The widely studied and available nickel has been the one of dominant membrane material for several decades. In this paper we introduce multilayer graphene as membrane material for a condenser microphone. The graphene device outperforms a high end commercial nickel-based microphone over a significant part of the acoustic spectrum, with a larger than 10 dB enhancement of sensitivity. Our experimental results are supported with numerical simulations, which show that a 300 layer thick graphene membrane under maximum tension would offer excellent extension of the frequency range, up to 1 MHz, with similar sensitivity as commercial condenser microphones.

Real-time audio processing of ambient sound

Abstract: An earpiece for real-time audio processing of ambient sound includes an ear bud that provides passive noise attenuation to the earpiece such that exterior ambient sound is substantially reduced within an ear of a wearer, an exterior microphone that receives ambient sound and converts the received ambient sound into analog electrical signals, and an analog-to-digital converter that converts the analog electrical signals into digital signals representative of the ambient sounds. The earpiece further includes a digital signal processor that performs a transformation operation on the digital signals according to instructions received from a mobile device, the transformation operation transforms the digital signals into modified digital signals, a digital-to-analog converter that converts the modified digital signals into modified analog electrical signals, and a speaker that outputs the modified analog electrical signals as audio waves.

Exploring the feasibility of smart phone microphone for measurement of acoustic voice parameters and voice pathology screening

Abstract: The objective of this study is to evaluate the reliability of acoustic voice parameters obtained using smart phone (SP) microphones and investigate the utility of use of SP voice recordings for voice screening. Voice samples of sustained vowel/a/obtained from 118 subjects (34 normal and 84 pathological voices) were recorded simultaneously through two microphones: oral AKG Perception 220 microphone and SP Samsung Galaxy Note3 microphone. Acoustic voice signal data were measured for fundamental frequency, jitter and shimmer, normalized noise energy (NNE), signal to noise ratio and harmonic to noise ratio using Dr. Speech software. Discriminant analysis-based Correct Classification Rate (CCR) and Random Forest Classifier (RFC) based Equal Error Rate (EER) were used to evaluate the feasibility of acoustic voice parameters classifying normal and pathological voice classes. Lithuanian version of Glottal Function Index (LT_GFI) questionnaire was utilized for self-assessment of the severity of voice disorder. The correlations of acoustic voice parameters obtained with two types of microphones were statistically significant and strong (r = 0.73–1.0) for the entire measurements. When classifying into normal/pathological voice classes, the Oral-NNE revealed the CCR of 73.7 % and the pair of SP-NNE and SP-shimmer parameters revealed CCR of 79.5 %. However, fusion of the results obtained from SP voice recordings and GFI data provided the CCR of 84.60 % and RFC revealed the EER of 7.9 %, respectively. In conclusion, measurements of acoustic voice parameters using SP microphone were shown to be reliable in clinical settings demonstrating high CCR and low EER when distinguishing normal and pathological voice classes, and validated the suitability of the SP microphone signal for the task of automatic voice analysis and screening.