Showing papers on "Microphone published in 2008"

Mobile tele-presence system with a microphone system

Abstract: A remote controlled robot system that includes a robot and a remote control station. The robot includes a binaural microphone system that is coupled to a speaker system of the remote control station. The binaural microphone system may include a pair of microphones located at opposite sides of a robot head. the location of the microphones roughly coincides with the location of ears on a human body. Such microphone location creates a mobile robot that more effectively simulates the tele-presence of an operator of the system. The robot may include two different microphone systems and the ability to switch between systems. For example, the robot may also include a zoom camera system and a directional microphone. The directional microphone may be utilized to capture sound from a direction that corresponds to an object zoomed upon by the camera system.

System and Method for Controlling the Operation of a Device by Voice Commands

Abstract: The present invention includes a speech recognition system comprising a light element, a power control switch, the power control switch varying the power delivered to the light element, a controller, a microphone, a speech recognizer coupled to the microphone for recognizing speech input signals and transmitting recognition results to the controller, and a speech synthesizer coupled to the controller for generating synthesized speech, wherein the controller varies the power to the light element in accordance with the recognition results received from the speech recognizer. Embodiments of the invention may alternatively include a low power wake up circuit. In another embodiment, the present invention is a method of controlling a device by voice commands.

The sources of jet noise: experimental evidence

Abstract: The primary objective of this investigation is to determine experimentally the sources of jet mixing noise. In the present study, four different approaches are used. It is reasonable to assume that the characteristics of the noise sources are imprinted on their radiation fields. Under this assumption, it becomes possible to analyse the characteristics of the far-field sound and then infer back to the characteristics of the sources. The first approach is to make use of the spectral and directional information measured by a single microphone in the far field. A detailed analysis of a large collection of far-field noise data has been carried out. The purpose is to identify special characteristics that can be linked directly to those of the sources. The second approach is to measure the coherence of the sound field using two microphones. The autocorrelations and cross-correlations of these measurements offer not only valuable information on the spatial structure of the noise field in the radial and polar angle directions, but also on the sources inside the jet. The third approach involves measuring the correlation between turbulence fluctuations inside a jet and the radiated noise in the far field. This is the most direct and unambiguous way of identifying the sources of jet noise. In the fourth approach, a mirror microphone is used to measure the noise source distribution along the lengths of high-speed jets. Features and trends observed in noise source strength distributions are expected to shed light on the source mechanisms. It will be shown that all four types of data indicate clearly the existence of two distinct noise sources in jets. One source of noise is the fine-scale turbulence and the other source is the large turbulence structures of the jet flow. Some of the salient features of the sound field associated with the two noise sources are reported in this paper.

Acoustic systems for electronic devices

Abstract: A portable electronic device may have acoustic ports such as microphone and speaker ports. Acoustic devices such as microphones and speakers may be associated with the acoustic ports. An acoustic port may have an opening between an interior and exterior of the portable electronic device. The opening may be covered by a metal mesh. An acoustic fabric may be interposed between the metal mesh and the opening. The opening may be formed from a hole in a glass member having outer and inner chamfers. A microphone boot may be provided that forms front and rear radial seals with a housing of the device and a microphone unit respectively. The microphone boot may also form multiple face seals with the microphone unit. A speaker for the speaker port may be enclosed in a sealed speaker enclosure. The speaker enclosure may have a pressure-equalizing vent slit covered with an acoustic mesh.

Digital Hearing Aids

Abstract: Preface Hearing-Aid Technology Types of Hearing Aids From Analog to Digital Digital Circuit Components Batteries Concluding Remarks References Signal Processing Basics Signal and System Properties Discrete Fourier Transform Filters and Filter Banks Block Processing Digital System Concerns Concluding Remarks References The Electroacoustic System Hearing Aid System Head and Ear Microphone and Receiver Vent Acoustics Occlusion Effect Concluding Remarks References Directional Microphones Hearing-Aid Microphones Directional Response Patterns Frequency Response Magnitude Frequency Response Microphone Mismatch Interaction with Vents Microphone Noise Microphones on the Head Microphone Performance Indices Rooms and Reverberation Benefit in the Real World Concluding Remarks References Adaptive and Multi-Microphone Arrays Two-Microphone Adaptive Array Delay-And-Sum Beamforming Adaptive Arrays Superdirective Arrays Widely-Spaced Arrays Array Benefits Concluding Remarks References Wind Noise Turbulence Hearing-Aid Measurements Signal Characteristics Wind-Noise Reduction Concluding Remarks References Feedback Cancellation The Feedback System Gain-Reduction Solutions Adaptive Feedback Cancellation Processing Limitations Concluding Remarks References Dynamic-Range Compression Does Compression Help? Algorithm Design Concerns Single-Channel Compression Multi-Channel Compression Frequency-Domain Compression Frequency Warping Concluding Remarks References Single-Microphone Noise Suppression Properties of Speech and Noise Signals Low-Level Expansion Envelope Valley Tracking Bandwidth Reduction Envelope Modulation Filters Concluding Remarks References Spectral Subtraction Noise Estimation Wiener Filter Spectral Subtraction Algorithm Effectiveness Concluding Remarks References Spectral Contrast Enhancement Auditory Filters in the Damaged Cochlea Spectral Valley Suppression Spectral Contrast Modification Excess Upward Spread of Masking F2/F1 Ratio Processing Comparison Combining Spectral Contrast Enhancement with Compression Concluding Remarks References Sound Classification The Rationale for Classification Signal Features Feature Selection Classifier Algorithms Classification Examples Concluding Remarks References Binaural Signal Processing The "Cocktail Party" Problem Signal Transmission Binaural Compression Binaural Noise Suppression Dichotic Band Splitting Concluding Remarks References Index

Automated sound recording and analysis techniques for bird surveys and conservation

Abstract: There is a great need for increased use and further development of automated sound recording and analysis of avian sounds. Birds are critical to ecosystem functioning so techniques to make avian monitoring more efficient and accurate will greatly benefit science and conservation efforts. We provide an overview of the hardware approaches to automated sound recording as well as an overview of the prominent techniques used in software to automatically detect and classify avian sound. We provide a comparative summary of examples of three general categories of hardware solutions for automating sound recording which include a hardware interface for a scheduling timer to control a standalone commercial recorder, a programmable recording device, and a single board computer. We also describe examples of the two main approaches to improving microphone performance for automated recorders through small arrays of microphone elements and using waveguides. For the purposes of thinking about automated sound analysis, we suggest five basic sound fragment types of avian sound and discuss a variety of techniques to automatically detect and classify avian sounds to species level, as well as their limitations. A variety of the features to measure for the various call types are provided, along with a variety of classification methods for those features. They are discussed in context of general performance as well as the monitoring and conservation efforts they are used in.

Method and device for sound detection and audio control

Abstract: At least one exemplary embodiment is directed to a listening device (100) can include a receiver (102) and means for directing a sound produced by the receiver into an ear of the user, a microphone (104) and means for mounting the microphone so as to receive the sound in an environment , detecting means for detecting an auditory signal in the sound received by the microphone, and alerting means for alerting the user to the presence of the auditory signal, whereby the user's personal safety is enhanced due to the user being alerted to the presence of the auditory signal, which otherwise may be unnoticed by the user due to loud sound level created at the ear of the user by the receiver.

Maximum Likelihood Sound Source Localization and Beamforming for Directional Microphone Arrays in Distributed Meetings

Abstract: In distributed meeting applications, microphone arrays have been widely used to capture superior speech sound and perform speaker localization through sound source localization (SSL) and beamforming. This paper presents a unified maximum likelihood framework of these two techniques, and demonstrates how such a framework can be adapted to create efficient SSL and beamforming algorithms for reverberant rooms and unknown directional patterns of microphones. The proposed method is closely related to steered response power-based algorithms, which are known to work extremely well in real-world environments. We demonstrate the effectiveness of the proposed method on challenging synthetic and real-world datasets, including over six hours of recorded meetings.

A Linear Closed-Form Algorithm for Source Localization From Time-Differences of Arrival

Abstract: Microphone arrays often operate in the near field, which complicates the problem of determining a source location from time-difference-of-arrival (TDOA) measurements typically derived from generalized cross-correlation functions. Each TDOA satisfies the equation of a hyperboloid in space and methods have been developed to either solve for intersecting hyperboloids or make some approximation to them, keeping source-location determination a nonlinear, somewhat complex problem. We introduce a closed-form, analytic solution for the problem (the GS algorithm). It is so simple that we were surprised that, until very recently, there have been no other solutions similar to ours. The method uses a minimum of five microphones in three dimensions, one more than other solutions, but, for nonsingular layouts of the microphones, it is very fast and accurate. First, the new method is compared to other closed-form methods for accuracy and sensitivity to noise using simulated data. Then, several variants of GS are compared to two other real-time algorithms, LEMSalg and SRP-PHAT, using real, human-talker data from a large array in a noisy room.

Systems, methods, and apparatus for multi-microphone based speech enhancement

Abstract: Systems, methods, and apparatus for processing an M-channel input signal are described that include outputting a signal produced by a selected one among a plurality of spatial separation filters. Applications to separating an acoustic signal from a noisy environment are described, and configurations that may be implemented on a multi-microphone handheld device are also described.

System and method for providing single microphone noise suppression fallback

Abstract: Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.

Ambient noise reduction system

Abstract: An adaptive, feed-forward, ambient noise-reduction system includes a reference microphone for generating first electrical signals representing incoming ambient noise, and a connection path including a circuit for inverting these signals and applying them to a loudspeaker directed into the ear of a user. The system also includes an error microphone for generating second electrical signals representative of sound (including that generated by the loudspeaker in response to the inverted first electrical signals) approaching the user's ear. An adaptive electronic filter is provided in the connection path, together with a controller for automatically adjusting one or more characteristics of the filter in response to the first and second electrical signals. The system is configured to constrain the operation of the adaptive filter such that it always conforms to one of a predetermined family of filter responses, thereby restricting the filter to operation within a predetermined and limited set of amplitude and phase characteristics.

Noise reduction device

Abstract: A noise reduction device, including a microphone for detecting noise, a noise controller for reversing the phase of noise detected by the microphone based on information outputted from the microphone, and a speaker for outputting sound based on information outputted from the noise controller, is arranged at a seat surrounded by a shell portion having a cavity, wherein a speaker is disposed in the cavity of the shell portion.

Multichannel acoustic echo reduction

Abstract: A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal.

Direct computation of sound and microphone locations from time-difference-of-arrival data

Abstract: In this paper we present a novel approach to directly recover the location of both microphones and sound sources from time-difference-of-arrival measurements only. No approximation solution is required for initialization and in the absence of noise our approach is guaranteed to always recover the exact solution. Our approach only requires solving linear equations and matrix factorization. We demonstrate the feasibility of our approach with synthetic data.

Method and device for audio recording

Abstract: At least one exemplary embodiment so directed to an earpiece (100). The earpiece can include an Ambient Sound Microphone (111) configured to capture ambient sound, an Ear Canal Microphone (123) configured to capture internal sound in the ear canal, a memory (208) configured to record at least a portion of the history of the ambient sound and the internal sound, and a processor (121) configured to save a recent portion of the history responsive to an event.

Multiple microphone voice activity detector

Abstract: Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

Always on headwear recording system

Abstract: A system that records audio and stores the recording is provided. The system includes first and second monitoring assemblies mounted in an earpiece that occludes and forms an acoustic seal of an ear canal. The first monitoring assembly includes an ambient sound microphone (ASM) to monitor an ambient acoustic field and produce an ASM signal. The second monitoring assembly includes an ear canal microphone (ECM) to monitor an acoustic field within the ear canal and produce an ECM signal. The system also includes a data storage device configured to act as a circular buffer for continually storing at least one of the ECM signal or the ASM signal, a further data storage device and a record-activation system. The record-activation system activates the further data storage device to record a content of the data storage device.

In-ear digital electronic noise cancelling and communication device

Abstract: A noise canceling and communication system is described. An in-ear device is adapted to fit in the ear canal of a device user. A passive noise reduction element reduces external noise entering the ear canal. An external microphone senses an external acoustic signal outside the ear canal to produce a representative external microphone signal. An internal microphone senses an internal acoustic signal proximal to the tympanic membrane to produce a representative internal microphone signal. One or more internal sound generators produce a noise cancellation signal and an acoustic communication signal, both directed towards the tympanic membrane. A probe tube shapes an acoustic response between the internal sound generator and the internal microphone to be relatively constant over a wide audio frequency band. An electronics module is located externally of the ear canal and in communication with the in-ear device for processing the microphone signals using a hybrid feed forward and feedback active noise reduction algorithm to produce the noise cancellation signal. The noise reduction algorithm includes a modeling component based on a transfer function associated with the internal sound generator and at least one of the microphones to automatically adjust the noise cancellation signal for fit and geometry of the ear canal of the user. The communication component also includes a modeling component based on a transfer function associated with the internal sound generator and at least one of the microphones to automatically adjust the communication signal for fit and geometry of the ear canal of the user and to assure that the communication signal does not interfere with the noise reduction algorithm and that the noise cancellation signal does not interfere with passing of the communication signal.

Microphone Array Shape Calibration in Diffuse Noise Fields

Abstract: This correspondence presents a microphone array shape calibration procedure for diffuse noise environments. The procedure estimates intermicrophone distances by fitting the measured noise coherence with its theoretical model and then estimates the array geometry using classical multidimensional scaling. The technique is validated on noise recordings from two office environments.

Beamforming pre-processing for speaker localization

Abstract: Embodiments of the present invention relate to methods, systems, and computer program products for signal processing. A first plurality of microphone signals is obtained by a first microphone array. A second plurality of microphone signals is obtained by a second microphone array different from the first microphone array. The first plurality of microphone signals is beamformed by a first beamformer comprising beamforming weights to obtain a first beamformed signal. The second plurality of microphone signals is beamformed by a second beamformer comprising the same beamforming weights as the first beamformer to obtain a second beamformed signal. The beamforming weights are adjusted such that the power density of echo components and/or noise components present in the first and second plurality of microphone signals is substantially reduced.

Side-ported MEMS microphone assembly

Abstract: A side-ported MEMS microphone package defines an acoustic path from a side of the package substrate to a microphone die disposed within a chamber defined by the substrate and a lid attached to the substrate. Optionally or alternatively, a circuit board, to which the microphone package is mounted, may define an acoustic path from an edge of the circuit board to a location under the microphone package, adjacent a bottom port on the microphone package. In either case, the acoustic path may be a hollow passage through at least a portion of the substrate or the circuit board. The passage may be defined by holes, channels, notches, etc. defined in each of several layers of a laminated substrate or circuit board, or the passage may be defined by holes drilled, molded or otherwise formed in a solid or laminated substrate or circuit board.

Method and Device for Voice Operated Control

Abstract: At least one exemplary embodiment is directed to a method and/or a device for voice operated control. The method can include method measuring an ambient sound received from at least one Ambient Sound Microphone, measuring an internal sound received from at least one Ear Canal Microphone, detecting a spoken voice from a wearer of the earpiece based on an analysis of the ambient sound and the internal sound, and controlling at least one voice operation of the earpiece if the presence of spoken voice is detected. The analysis can be a non-difference comparison such as a correlation analysis, a cross-correlation analysis, and a coherence analysis.

Audio camera using microphone arrays for real time capture of audio images and method for jointly processing the audio images with video images

Abstract: A method comprises providing at least one processing unit comprising a decomposing section and a playback section; receiving, at the decomposing section, audio data generated via an array of microphones, the audio data representing an acoustic scene; decomposing the audio data into a plurality of signals representing components of the acoustic scene arriving from a plurality of directions, using the decomposing section; and rendering the audio components for a listener based on the plurality of directions of the audio components, using the playback section.

Multifunction System and Method for Integrated Hearing and Communication with Noise Cancellation and Feedback Management

Abstract: Systems, devices and methods for communication include an ear canal microphone configured for placement in the ear canal to detect high frequency sound localization cues. An external microphone positioned away from the ear canal can detect low frequency sound, such that feedback can be substantially reduced. The canal microphone and the external microphone are coupled to a transducer, such that the user perceives sound from the external microphone and the canal microphone with high frequency localization cues and decreased feedback. Wireless circuitry can be configured to connect to many devices with a wireless protocol, such that the user can receive and transmit audio signals. A bone conduction sensor can detect near-end speech of the user for transmission with the wireless circuitry in noisy environment. Noise cancellation of background sounds near the user can improve the user's hearing of desired sounds.

Compressive wireless arrays for bearing estimation

Abstract: Joint processing of sensor array outputs improves the performance of parameter estimation and hypothesis testing problems beyond the sum of the individual sensor processing results. When the sensors have high data sampling rates, arrays are tethered, creating a disadvantage for their deployment and also limiting their aperture size. In this paper, we develop the signal processing algorithms for randomly deployable wireless sensor arrays that are severely constrained in communication bandwidth. We focus on the acoustic bearing estimation problem and show that when the target bearings are modeled as a sparse vector in the angle space, low dimensional random projections of the microphone signals can be used to determine multiple source bearings by solving an l 1-norm minimization problem. Field data results are shown where only 10 bits of information is passed from each microphone to estimate multiple target bearings.

Joint Dereverberation and Residual Echo Suppression of Speech Signals in Noisy Environments

Abstract: Hands-free devices are often used in a noisy and reverberant environment. Therefore, the received microphone signal does not only contain the desired near-end speech signal but also interferences such as room reverberation that is caused by the near-end source, background noise and a far-end echo signal that results from the acoustic coupling between the loudspeaker and the microphone. These interferences degrade the fidelity and intelligibility of near-end speech. In the last two decades, post filters have been developed that can be used in conjunction with a single microphone acoustic echo canceller to enhance the near-end speech. In previous works, spectral enhancement techniques have been used to suppress residual echo and background noise for single microphone acoustic echo cancellers. However, dereverberation of the near-end speech was not addressed in this context. Recently, practically feasible spectral enhancement techniques to suppress reverberation have emerged. In this paper, we derive a novel spectral variance estimator for the late reverberation of the near-end speech. Residual echo will be present at the output of the acoustic echo canceller when the acoustic echo path cannot be completely modeled by the adaptive filter. A spectral variance estimator for the so-called late residual echo that results from the deficient length of the adaptive filter is derived. Both estimators are based on a statistical reverberation model. The model parameters depend on the reverberation time of the room, which can be obtained using the estimated acoustic echo path. A novel postfilter is developed which suppresses late reverberation of the near-end speech, residual echo and background noise, and maintains a constant residual background noise level. Experimental results demonstrate the beneficial use of the developed system for reducing reverberation, residual echo, and background noise.

Piezoelectric microphone built on circular diaphragm

Abstract: This paper describes a piezoelectric microphone built on a circular diaphragm (CD), which is fabricated using the boron etching stop method. ZnO was used as the piezoelectric material, and the diaphragm material is low-stress SixNy. The diameter and thickness of the circular diaphragm are 2 mm and 1m, respectively. The thickness of the boron-doped layer – a support layer for the circular diaphragm – is approximately 7.4m. Based on ANSYS simulations, the sensitivity increment of the CD microphone was comparable to that of a square diaphragm (SD) microphone. The sensitivity of the CD microphone increased to 197% of that of a SD microphone, as expected from the simulation results. The first resonance frequency of the CD microphone is 54.8 kHz. At 1 kHz, the displacement of the CD microphone and SD microphone are 3.96 nm and 1.54 nm, respectively. The fabricated CD microphone was connected to an amplifier system to confirm the ability to reproduce a human voice. The gain of the amplifier system is 200. © 2008 Elsevier B.V. All rights reserved.

Noise reduction apparatus and audio reproduction apparatus

Abstract: A noise reduction apparatus includes: a speaker with a speaker unit held by holding means to make it possible to mix sounds emitted from front and rear of a vibration plate of the speaker; a microphone provided in an area where the sounds emitted from the front and rear of the vibration plate of the speaker are mixed and cancelled; and means for supplying a noise reduction audio signal obtained by phase-inverting an audio signal collected by the microphone to the speaker.

The Spherical-Shell Microphone Array

Abstract: Spherical microphone arrays have been recently studied for a wide range of applications. In particular, microphones arranged around an open or virtual sphere are useful in scanning microphone arrays for sound field analysis. However, open-sphere spherical arrays have been shown to have poor robustness at frequencies related to the zeros of the spherical Bessel functions. This paper presents a framework for the analysis of array robustness using the condition number of a given matrix, and then proposes several robust array configurations. In particular, a dual-sphere configuration previously presented which uses twice as many microphones compared to a single-sphere configuration is analyzed. This paper then shows that high robustness can be achieved without increasing the number of microphones by arranging the microphones in the volume of a spherical shell. Another simpler configuration employs a single sphere and an additional microphone at the sphere center, showing improved robustness at the low-frequency range. Finally, the white-noise gain of the arrays is investigated verifying that improved white-noise gain is associated with lower matrix condition number.