Showing papers on "Microphone published in 2010"

Method and system for presence detection

Abstract: The transmission system transmits an encrypted identifier ultrasonically in an open-air environment using an ultrasonic transmitter. The ultrasonic identifier is associated with a location, which may be a store or a department within a store, and may be received by a microphone on a mobile phone. The identifier may be used to infer presence of the mobile phone user at the location associated with the identifier. The transmitter may include an ultrasonic transducer. The system may further provide a reward, responsive to inferring presence at the location.

Speakerphone and/or microphone arrays and methods and systems of using the same

Abstract: The present disclosure is directed to devices, methods and systems for microphone arrays wherein enhancing performance of directional microphone arrays is provided. Enhanced performance of speaker phones is also provided. In certain embodiments, the housing of the device is configured to support the at least three microphones and the loudspeaker in a substantially first orientation; and the at least three microphones and the loudspeaker are arranged in a spatial relationship such that appropriate phase and delay characteristics achieve a substantial null response in the at least three microphones and in the loudspeaker in a substantial vertical direction away from the substantially first orientation over a desired audible range of frequencies and the device is able to provide a response to sounds over a range of first oriented elevations.

Acoustic side-channel attacks on printers

Abstract: We examine the problemof acoustic emanations of printers We present a novel attack that recovers what a dot-matrix printer processing English text is printing based on a record of the sound it makes, if the microphone is close enough to the printer In our experiments, the attack recovers up to 72 % of printed words, and up to 95 % if we assume contextual knowledge about the text, with a microphone at a distance of 10cmfrom the printer After an upfront training phase, the attack is fully automated and uses a combination of machine learning, audio processing, and speech recognition techniques, including spectrum features, Hidden Markov Models and linear classification; moreover, it allows for feedback-based incremental learning We evaluate the effectiveness of countermeasures, and we describe how we successfully mounted the attack in-field (with appropriate privacy protections) in a doctor's practice to recover the content of medical prescriptions

Personal acoustic device position determination

Abstract: A apparatus and method for determining an operating state of an earpiece of a personal acoustic device and/or the entirety of the personal acoustic device by analyzing signals output by at least an inner microphone disposed within a cavity of a casing of the earpiece and an outer microphone disposed on the personal acoustic device in a manner acoustically coupling it to the environment outside the casing of the earpiece.

Guidelines for Selecting Microphones for Human Voice Production Research

Abstract: Purpose This tutorial addresses fundamental characteristics of microphones (frequency response, frequency range, dynamic range, and directionality), which are important for accurate measurements of voice and speech. Method Technical and voice literature was reviewed and analyzed. The following recommendations on desirable microphone characteristics were formulated: The frequency response of microphones should be flat (i.e., variation of less than 2 dB) within the frequency range between the lowest expected fundamental frequency of voice and the highest spectral component of interest. The equivalent noise level of the microphones is recommended to be at least 15 dB lower than the sound level of the softest phonations. The upper limit of the dynamic range of the microphone should be above the sound level of the loudest phonations. Directional microphones should be placed at the distance that corresponds to their maximally flat frequency response, to avoid the proximity effect; otherwise, they will be unsuit...

An i-vector extractor suitable for speaker recognition with both microphone and telephone speech

Abstract: It is widely believed that speaker verification systems perform better when there is sufficient background training data to deal with nuisance effects of transmission channels. It is also known that these systems perform at their best when the sound environment of the training data is similar to that of the context of use (test context). For some applications however, training data from the same type of sound environment is scarce, whereas a considerable amount of data from a different type of environment is available. In this paper, we propose a new architecture for text-independent speaker verification systems that are satisfactorily trained by virtue of a limited amount of application-specific data, supplemented with a sufficient amount of training data from some other context. This architecture is based on the extraction of parameters (i-vectors) from a low-dimensional space (total variability space) proposed by Dehak [1]. Our aim is to extend Dehak’s work to speaker recognition on sparse data, namely microphone speech. The main challenge is to overcome the fact that insufficient application-specific data is available to accurately estimate the total variability covariance matrix. We propose a method based on Joint Factor Analysis (JFA) to estimate microphone eigenchannels (sparse data) with telephone eigenchannels (sufficient data). For classification, we experimented with the following two approaches: Support Vector Machines (SVM) and Cosine Distance Scoring (CDS) classifier, based on cosine distances. We present recognition results for the part of female voices in the interview data of the NIST 2008 SRE. The best performance is obtained when our system is fused with the state-of-the-art JFA. We achieve 13% relative improvement on equal error rate and the minimum value of detection cost function decreases from 0.0219 to 0.0164.

Object-based activity recognition with heterogeneous sensors on wrist

Abstract: This paper describes how we recognize activities of daily living (ADLs) with our designed sensor device, which is equipped with heterogeneous sensors such as a camera, a microphone, and an accelerometer and attached to a user's wrist Specifically, capturing a space around the user's hand by employing the camera on the wrist mounted device enables us to recognize ADLs that involve the manual use of objects such as making tea or coffee and watering plant Existing wearable sensor devices equipped only with a microphone and an accelerometer cannot recognize these ADLs without object embedded sensors We also propose an ADL recognition method that takes privacy issues into account because the camera and microphone can capture aspects of a user's private life We confirmed experimentally that the incorporation of a camera could significantly improve the accuracy of ADL recognition

Method and apparatus for voice controlled operation of a media player

Abstract: A system and methods for voice controlled operation of a media player are provided. In one embodiment, a method includes detecting user positioning of a microphone power switch to an off position, detecting user positioning of the microphone power switch to an on position within a predetermined period of time and entering a voice recognition mode, by the media player, based on the user positioning of the microphone power switch to the on position within the predetermined period of time. The method may further include detecting one or more output signals of the microphone, detecting a voice command based on the one or more output signals of the microphone, and controlling operation of the media player based on the voice command, wherein the media player outputs a graphical display associated with the voice command.

Noise profile determination for voice-related feature

Abstract: Systems, methods, and devices for noise profile determination for a voice-related feature of an electronic device are provided. In one example, an electronic device capable of such noise profile determination may include a microphone and data processing circuitry. When a voice-related feature of the electronic device is not in use, the microphone may obtain ambient sounds. The data processing circuitry may determine a noise profile based at least in part on the obtained ambient sounds. The noise profile may enable the data processing circuitry to at least partially filter other ambient sounds obtained when the voice-related feature of the electronic device is in use.

A Two Microphone-Based Approach for Source Localization of Multiple Speech Sources

Abstract: This paper proposes a two microphone-based source localization technique for multiple speech sources utilizing speech specific properties and novel clustering algorithms. Voiced speech is sparse in the frequency domain and can be represented by sinusoidal tracks via sinusoidal modeling which provides high local signal-to-noise ratio (SNR). By utilizing the inter-channel phase differences (IPDs) between the dual channels on the sinusoidal tracks, the source localization of the mixed multiple speech sources is turned into a clustering problem on the IPD versus frequency plot. The generalized mixture decomposition algorithm (GMDA) is used to cluster the groups of points corresponding to multiple sources and thus estimate the direction of arrival (DOA) of the sources. Experiments illustrate the proposed GMDA algorithm with the Laplacian noise model can estimate the number of sources accurately and exhibits smaller DOA estimation error than the baseline histogram based DOA estimation algorithm in various scenarios including reverberant and additive white noise environments. Experiments suggest that appropriate power thresholding can be a simple and good approximation to the sinusoidal modeling, for the purpose of selecting time-frequency points with high local SNR, with slight loss in performance.

Rotating Blade Trailing-Edge Noise: Experimental Validation of Analytical Model

Abstract: This paper deals with the experimental validation of an analytical trailing-edge noise model dedicated to low-speed fans operating in free field. The model is intrinsically related to the aerodynamics of the blades and should lead to a useful fast-running tool to be included in a blade-design process in an industrial context. The investigations are made on a two-bladed low-speed axial fan without shroud, installed inside an anechoic room. The blades are instrumented with two sets of embedded small-size microphones (2.5 mm diam), and the wall-pressure signals are acquired via a slip ring mounted on the fan axis. The chord-based Reynolds number is about 200,000, and the tip Mach number about 0.07. The data base is completed by far-field measurements made with a single microphone on a moving support. The analytical model is based on a previously published extension of Amiet's trailing-edge noise theory. A blade is split into several strips in the spanwise direction, and the model is applied to each strip. For this the input data are interpolated from the measurements performed with the aforementioned sets of microphones. The trailing-edge noise model is more reliable for observer positions within ±30° from the fan-rotation plane.

Room acoustics simulation for multichannel microphone arrays

Abstract: A new and efficient room acoustics simulation software package for MATLAB is presented which can simulate recordings of arbitrary microphone arrays within an echoic room. This simulator supports research related to developing and experimenting with multichannel microphone arrays and higher order ambisonic playback. Based upon the work by Schimmel et al. [1], this new simulation software package models both specular and diffuse reflections in a shoebox type environment. It is an improvement over previous work as it can simulate microphone arrays with arbitrary directional impulse responses and large numbers of sensors. The spherical harmonic expansion up to a specified order at any point within an echoic room can also be simulated by specifying a microphone array with custom directional gains that match the spherical harmonic functions. Furthermore, this simulator provides realistic phase information for the sound recorded by the microphone array, including accurate inter-sensor time delays for the sources and early specular reflections. The room acoustics simulator is implemented as a C program that interfaces with MATLAB and is freely available from the authors.

Spherical Microphone Array Beamforming

Abstract: Spherical microphone arrays have been recently studied for spatial sound recording, speech communication, and sound field analysis for room acoustics and noise control. Complementary studies presented progress in beamforming methods. This chapter reviews beamforming methods recently developed for spherical arrays, from the widely used delay-and-sum and Dolph-Chebyshev, to the more advanced optimal methods, typically performed in the spherical harmonics domain.

Accessory controller for electronic devices

Abstract: Accessories such as headsets for electronic devices are provided. A headset may be provided with a button controller assembly that has user-actuated buttons and a microphone. The microphone may be formed by mounting a microphone transducer on a printed circuit board. A housing may be mounted over the transducer to form a sealed cavity for the transducer. Circuitry may be mounted on portions of the printed circuit board that extend beyond the edges of the microphone housing. The button controller assembly may have dome switches. The dome switches may have a housing that encloses dome switch components and that forms a structural internal part for the button controller. The dome switch housing structure may have tabs or other engagement features that mate with corresponding engagement features in a button member. The button member may be pressed by a user to actuate a desired dome switch.

Adaptive noise cancellation

Abstract: In some embodiments a noise cancellation system includes a first digital microphone to detect ambient noise, a first sigma delta modulator coupled to an output of the first digital microphone, a second digital microphone located near an earpiece speaker to detect an output of the earpiece speaker, a second sigma delta modulator coupled to an output of the second digital microphone, a decimator coupled to the second sigma delta modulator, and an adaptive digital filter to adaptively adjust an output of the earpiece speaker in response to the decimator and the first sigma delta modulator so that the output of the earpiece speaker includes a desired audio and an acoustic signal to cancel some or all of the ambient noise Other embodiments are described and claimed

3D source localization in the spherical harmonic domain using a pseudointensity vector

Abstract: The problem of acoustic source localization is important in many acoustic signal processing applications, such as distant speech acquisition and automated camera steering. In noisy and reverberant environments, the source localization problem becomes challenging and many existing algorithms deteriorate. Three-dimensional source localization presents advantages for certain applications such as beamforming, where we can steer a beam to both the desired azimuth and the desired elevation. In this paper, we present an acoustic source localization method with low computational complexity which, instead of using individual microphone signals, combines them to form eigenbeams. We then use the zero-and first-order eigenbeams to compute a pseudointensity vector pointing in the direction of the sound source. In an experimental study, the proposed method's localization accuracy is compared with that of a steered response power localization method, which uses the same eigenbeams. The results demonstrate that the proposed method has higher localization accuracy.

Wireless multi-user audio system

Abstract: Various methods and devices are provided for a wireless audio system for a number of users. The system includes a base unit that is adapted to removably store, recharge and communicate with various communication modules, including personal microphone modules, table-top microphones, and audio adapters. The system also includes a plurality of personal microphone modules that are each adapted to be removable and coupled, for example, to a user's clothing, and further, are adapted to communicate wirelessly with the base unit, and table-top microphones that are adapted to communicate wirelessly with the base unit.

Database of omnidirectional and B-format room impulse responses

Abstract: This paper introduces a new database of room impulse responses. This database differs greatly from previously released databases as it contains over 700 impulse responses. The impulse responses are measured in three different rooms each with a static source position and at least 130 different receiver positions. Each measurement position is recorded with both an omnidirectional microphone and a B-format microphone.

Interactive video system

Abstract: An interactive video system is provided that is capable of combining streaming televised events with video conferencing technology to create a social television experience. A first user is able to connect via a webcam and microphone to a server that combines the webcam video and sound from the microphone with a streaming video that could be a televised event. The combined webcam video, microphone sound, and streaming video is then broadcast to the first user and other users such that the users are capable of viewing the streaming video with the live conferencing video and sound from the first user. This procedure may be repeated such that a plurality of users may video conference live on top of the streaming video.

Active noise cancellation

Abstract: This document discusses, among other things, systems and methods for active noise cancellation. One example system includes a digital ANC circuit configured to receive first audio information from a first microphone and to produce an a digital anti-noise signal configured to attenuate noise sensed by the first microphone; an analog ANC circuit configured to receive second audio information from a second microphone and to produce an analog anti-noise signal configured to attenuate noise sensed by the second microphone; and wherein the system is configured to receive an intended audio signal and to provide an output signal for a speaker using the intended audio signal, the analog anti-noise signal, and the digital anti-noise signal.

Beamforming with a circular microphone array for localization of environmental noise sources

Abstract: It is often enough to localize environmental sources of noise from different directions in a plane. This can be accomplished with a circular microphone array, which can be designed to have practically the same resolution over 360°. The microphones can be suspended in free space or they can be mounted on a solid cylinder. This investigation examines and compares two techniques based on such arrays, the classical delay-and-sum beamforming and an alternative method called circular harmonics beamforming. The latter is based on decomposing the sound field into a series of circular harmonics. The performance of the two signal processing techniques is examined using computer simulations, and the results are validated experimentally.

Sound monitoring system and speech collection system

Abstract: Monitoring accuracy degrades due to a noise in an environment where there are many sound sources other than those to be monitored. Easy initialization is required for an environment where many apparatuses operate. A sound monitoring system includes a microphone array having multiple microphones and a location-based abnormal sound monitoring section as a processing section. The location-based abnormal sound monitoring section is supplied with an input signal from the microphone array via a waveform acquisition section and a network. Using the input signal, the location-based abnormal sound monitoring section detects a temporal change in a sound source direction histogram. Based on a detected change result, the location-based abnormal sound monitoring section checks for abnormality in a sound field and outputs a monitoring result. The processing section searches for a microphone array near the sound source to be monitored. The processing section selects a sound field monitoring function for the sound source to be monitored based on various data concerning a microphone belonging to the searched microphone array.

A covariance fitting approach for correlated acoustic source mapping.

Abstract: Microphone arrays are commonly used for noise source localization and power estimation in aeroacoustic measurements. The delay-and-sum (DAS) beamformer, which is the most widely used beamforming algorithm in practice, suffers from low resolution and high sidelobe level problems. Therefore, deconvolution approaches, such as the deconvolution approach for the mapping of acoustic sources (DAMAS), are often used for extracting the actual source powers from the contaminated DAS results. However, most deconvolution approaches assume that the sources are uncorrelated. Although deconvolution algorithms that can deal with correlated sources, such as DAMAS for correlated sources, do exist, these algorithms are computationally impractical even for small scanning grid sizes. This paper presents a covariance fitting approach for the mapping of acoustic correlated sources (MACS), which can work with uncorrelated, partially correlated or even coherent sources with a reasonably low computational complexity. MACS minimizes a quadratic cost function in a cyclic manner by making use of convex optimization and sparsity, and is guaranteed to converge at least locally. Simulations and experimental data acquired at the University of Florida Aeroacoustic Flow Facility with a 63-element logarithmic spiral microphone array in the absence of flow are used to demonstrate the performance of MACS.

Plane-Wave Decomposition of Acoustical Scenes Via Spherical and Cylindrical Microphone Arrays

Abstract: Spherical and cylindrical microphone arrays offer a number of attractive properties such as direction-independent acoustic behavior and ability to reconstruct the sound field in the vicinity of the array. Beamforming and scene analysis for such arrays is typically done using sound field representation in terms of orthogonal basis functions (spherical/cylindrical harmonics). In this paper, an alternative sound field representation in terms of plane waves is described, and a method for estimating it directly from measurements at microphones is proposed. It is shown that representing a field as a collection of plane waves arriving from various directions simplifies source localization, beamforming, and spatial audio playback. A comparison of the new method with the well-known spherical harmonics based beamforming algorithm is done, and it is shown that both algorithms can be expressed in the same framework but with weights computed differently. It is also shown that the proposed method can be extended to cylindrical arrays. A number of features important for the design and operation of spherical microphone arrays in real applications are revealed. Results indicate that it is possible to reconstruct the sound scene up to order p with p2 microphones spherical array.

Sound Propagation: An Impedance Based Approach

Abstract: In a previous publication by the authors, a virtual measurement environment for sound-source localization on vibrating structures was presented. Based on surface velocity data obtained from Laser-Scanning-Vibrometry measurements, the Boundary-Element-Method (BEM) is used to simulate the sound radiation from a vibrating plate towards a microphone array under ideal conditions. The advantage of this approach is that the measurement conditions can be perfectly controlled and real sources can be considered, without restrictions on the type of source. The virtual measurement environment will now be used to investigate the effect of some of the uncertainties that can be encountered during beamforming measurements. For the most common planar array geometries, the beamforming source maps will be calculated for varying Signal-to-Noise Ratios (SNR) and different array imperfections (uncertainties in the microphone locations and deviation from the omni-directional directivity pattern of the microphones). As a measure of comparison, the two-dimensional normalized cross-correlation coefficient between the ideal source map and the source map with added uncertainties will be evaluated and discussed.

Method and arrangement for noise cancellation in a speech encoder

Abstract: The present invention relates to a method and arrangement for an improved noise canceller in a speech encoder. Sound signals are captured at a primary microphone in conjunction with a reference microphone. An adaptive shadow filter is adapted to the correlation between the signals captured at the primary and reference microphones. Further, a diffuse-noise-field detector is introduced which detects the presence of diffuse noise. When the diffuse-noise-field detector detects diffuse noise, the filter coefficients of the adapted shadow filter is used by a primary filter to cancel the diffuse noise at the signal captured by the primary microphone. Since the filter coefficients of the adapted shadow filter only is used for cancellation when diffuse noise is solely detected, cancellation of the speech signal is avoided.

Geometric reconstruction of the environment from its response to multiple acoustic emissions

Abstract: In this paper we propose a method for reconstructing the 2D geometry of the surrounding environment based on the signals acquired by a fixed microphone, when a series of acoustic stimula are produced in different positions in space. After estimating the Times Of Arrival (TOAs) of the reflective paths, we turn each TOA into a projective geometric constraint that can be used for determining the locations of the reflectors. The result consists of a collection of planar surfaces that correspond to the reflectors' locations. In this paper we present the whole processing chain and prove its effectiveness through experimental results.

Using phased array beamforming to identify broadband noise sources in a turbofan engine

Abstract: Fan broadband noise is a major component of the total noise emitted by turbofan engines, especially at lower shaft speeds. It is generated in the rotor/stator region, but the exact origin is not always known. This article discusses the application of phased array beamforming techniques for a better understanding of the source mechanisms of fan broadband noise. The Conventional Beamforming technique was applied, as well as the deconvolution technique CLEAN-SC and the beamforming technique ROSI for rotating sources. Beamforming was applied to acoustic data measured by two circular microphone arrays that were mounted in the intake and in the bypass of a Rolls-Royce fan rig. These arrays are normally used for the detection of azimuthal modes. The merits of beamforming are discussed by considering a number of typical low shaft speed cases. Using the intake array, in one of the cases forward radiating broadband noise sources were found that were coherent over a large area. These could have been due to a rotor i...