Showing papers on "Microphone array published in 2006"

Instability waves in a subsonic round jet detected using a near-field phased microphone array

Abstract: We propose a diagnostic technique to detect instability waves in a subsonic round jet using a phased microphone array. The detection algorithm is analogous to the beam-forming technique, which is typically used with a far-field microphone array to localize noise sources. By replacing the reference solutions used in the conventional beam-forming with eigenfunctions from linear stability analysis, the amplitudes of instability waves in the axisymmetric and first two azimuthal modes are inferred. Experimental measurements with particle image velocimetry and a database from direct numerical simulation are incorporated to design a conical array that is placed just outside the mixing layer near the nozzle exit. The proposed diagnostic technique is tested in experiments by checking for consistency of the radial decay, streamwise evolution and phase correlation of hydrodynamic pressure. The results demonstrate that in a statistical sense, the pressure field is consistent with instability waves evolving in the turbulent mean flow from the nozzle exit to the end of the potential core, particularly near the most amplified frequency of each azimuthal mode. We apply this technique to study the effects of jet Mach number and temperature ratio on the azimuthal mode balance and evolution of instability waves. We also compare the results from the beam-forming algorithm with the proper orthogonal decomposition and discuss some implications for jet noise.

Methods and apparatus for targeted sound detection

Abstract: Targeted sound detection methods and apparatus are disclosed. A microphone array has two or more microphones M0 . . . MM. Each microphone is coupled to a plurality of filters. The filters are configured to filter input signals corresponding to sounds detected by the microphones thereby generating a filtered output. One or more sets of filter parameters for the plurality of filters are pre-calibrated to determine one or more corresponding pre-calibrated listening zones. Each set of filter parameters is selected to detect portions of the input signals corresponding to sounds originating within a given listening zone and filter out sounds originating outside the given listening zone. A particular pre-calibrated listening zone is selected at a runtime by applying to the plurality of filters a set of filter coefficients corresponding to the particular pre-calibrated listening zone. As a result, the microphone array may detect sounds originating within the particular listening sector and filter out sounds originating outside the particular listening zone.

Superdirective Beamforming Robust Against Microphone Mismatch

Abstract: Fixed superdirective beamformers using small-size microphone arrays are known to be highly sensitive to errors in the assumed microphone array characteristics. This paper discusses the design of robust superdirective beamformers by taking into account the statistics of the microphone characteristics. Different design procedures are considered: applying a white noise gain constraint, trading off the mean noise and distortion energy, and maximizing the mean or the minimum directivity factor. When computational complexity is not important, maximizing the mean or the minimum directivity factor is the preferred design procedure. In addition, it is shown how to determine a suitable parameter range for the other design procedures.

Robust 3D Localization and Tracking of Sound Sources Using Beamforming and Particle Filtering

Abstract: In this paper we present a new robust sound source localization and tracking method using an array of eight microphones (US patent pending). The method uses a steered beamformer based on the reliability-weighted phase transform (RWPHAT) along with a particle filter-based tracking algorithm. The proposed system is able to estimate both the direction and the distance of the sources. In a videoconferencing context, the direction was estimated with an accuracy better than one degree while the distance was accurate within 10% RMS. Tracking of up to three simultaneous moving speakers is demonstrated in a noisy environment.

Ultra small microphone array

Abstract: Methods and apparatus for signal processing are disclosed. A discrete time domain input signal x m (t) may be produced from an array of microphones M 0 . . . M M . A listening direction may be determined for the microphone array. The listening direction is used in a semi-blind source separation to select the finite impulse response filter coefficients b 0 , b 1 . . . , b N to separate out different sound sources from input signal x m (t). One or more fractional delays may optionally be applied to selected input signals x m (t) other than an input signal x 0 (t) from a reference microphone M 0 . Each fractional delay may be selected to optimize a signal to noise ratio of a discrete time domain output signal y(t) from the microphone array. The fractional delays may be selected to such that a signal from the reference microphone M 0 is first in time relative to signals from the other microphone(s) of the array. A fractional time delay Δ may optionally be introduced into an output signal y(t) so that: y(t+Δ)=x(t+Δ)*b 0 +x(t−1+Δ)*b 1 +x(t−2+Δ)*b 2 + . . . +x(t−N+Δ)b N , where Δ is between zero and ±1.

Multiple Sound Source Mapping for a Mobile Robot by Self-motion Triangulation

Abstract: The paper describes a 2D sound source mapping system for a mobile robot. We developed a multiple sound sources localization method for a mobile robot with a 32 channel concentric microphone array. The system can separate multiple moving sound sources using direction localization. Directional localization and separation of different pressure sound sources is achieved using the delay and sum beam forming (DSBF) and the frequency band selection (FBS) algorithm. Sound sources were mapped by using a wheeled robot equipped with the microphone array. The robot localizes sounds direction on the move and estimates sound sources position using triangulation. Assuming the movement of sound sources, the system set a time limit and uses only the last few seconds data. By using the random sample consensus (RANSAC) algorithm for position estimation, we achieved 2D multiple sound source mapping from time limited data with high accuracy. Also, moving sound source separation is experimentally demonstrated with segments of the DSBF enhanced signal derived from the localization process

Methods and apparatuses for adjusting a listening area for capturing sounds

Abstract: In one embodiment, the methods and apparatuses adjust a listening area of a microphone includes detecting an initial listening zone; capture a captured sound through a microphone array; identify an initial sound based on the captured sound and the initial listening zone wherein the initial sound includes sounds within the initial listening zone; adjust the initial listening zone and forming the adjusted listening zone; and identify an adjusted sound based on the captured sound and the adjusted listening zone wherein the adjusted sound includes sounds within the adjusted listening zone.

Forming beams with nulls directed at noise sources

Abstract: A communication system (e.g., a speakerphone) includes an array of microphones, a speaker, memory and a processor. The processor may perform a virtual broadside scan on the microphone array and analyze the resulting amplitude envelope to identify acoustic source angles. Each of the source angles may be further investigated with a directed beam (e.g., a hybrid superdirective/delay-and-sum beam) to obtain a corresponding beam signal. Each source may be classified as either intelligence or noise based on an analysis of the corresponding beam signal. The processor may design a virtual beam pointed at an intelligence source and having nulls directed at one or more of the noise sources. Thus, the virtual beam may be highly sensitive to the intelligence source and insensitive to the noise sources.

Dynamic binaural sound capture and reproduction in focused or frontal applications

Abstract: A new approach to capturing and reproducing either live or recorded three-dimensional sound is described. Called MTB2.5 for “Motion-Tracked Binaural 2.5,” the method employs six real or simulated microphones, a head tracker, and special signal-processing procedures to combine the signals picked up by the microphones. In the microphone array, two central microphones are placed at the positions of the ears of a listener facing forward, and four peripheral microphones are located in pairs on either side of the central microphones. Only the low-frequency components of the four peripheral microphone signals are needed, and the total bandwidth required is approximately 2.5 times the bandwidth of one full-bandwidth audio channel. MTB2.5 achieves a high degree of realism by effectively placing the listener's ears in the space where the sounds are occurring, moving the virtual ears in synchrony with the listener's head motions. MTB2.5 also provides a universal format for recording spatial sound.

An automated acoustic systemto monitor and classify birds

Abstract: This paper presents a novel bird monitoring and recognition system in noisy environments. The project objective is to avoid bird strikes to aircraft. First, a cost-effective microphone dish concept (microphone array with many concentric rings) is presented that can provide directional and accurate acquisition of bird sounds and can simultaneously pick up bird sounds from different directions. Second, direction-of-arrival (DOA) and beamforming algorithms have been developed for the circular array. Third, an efficient recognition algorithm is proposed which uses Gaussian mixture models (GMMs). The overall system is suitable for monitoring and recognition for a large number of birds. Fourth, a hardware prototype has been built and initial experiments demonstrated that the array can acquire and classify birds accurately.

Post filter for microphone array

Abstract: The post filter for microphone array comprises a microphone array (10) consisting of at least two microphones for inputting a voice signal, a beam forming unit (13) for forming the voice signal inputted from the microphone array, a divider (14) for dividing a target sound including noise inputted from the microphone array into at last two frequency bands with a predetermined frequency, a first filter (20) for estimating the filter gain when the noise is uncorrelated between the microphones, a second filter (30) for estimating the filter gain of an average signal of one microphone in the microphone array or the microphone array, an adder (40) for adding the outputs from the first and second filters, and a means (41) for reducing noise based on the outputs from the adder and the beam forming unit.

Equalization in an extended area using multichannel inversion and wave field synthesis

Abstract: Wave field synthesis (WFS) targets the synthesis of the physical characteristics of a sound field in an extended listening area. This synthesis is, however, accompanied by noticeable reconstruction artifacts. They are due to both loudspeaker radiation characteristics and approximations to the underlying physical principles. These artifacts may introduce coloration, which must be.compensated for over the entire listening area. Multichannel equalization techniques allow for the control of the sound field produced by a loudspeaker array at a limited number of positions. The control can be extended to a large portion of space by employing a new method that combines multichannel equalization with a linear microphone array-based description of the sound field and accounts for WFS rendering characteristics and limitations. The proposed method is evaluated using an objective coloration criterion. Its benefits compared to conventional equalization techniques are pointed out for both ideal omnidirectional loudspeakers and multi-actuator panels.

Microphone array speaker localizers using spatial-temporal information

Abstract: A dual-step approach for speaker localization based on a microphone array is addressed in this paper. In the first stage, which is not the main concern of this paper, the time difference between arrivals of the speech signal at each pair of microphones is estimated. These readings are combined in the second stage to obtain the source location. In this paper, we focus on the second stage of the localization task. In this contribution, we propose to exploit the speaker's smooth trajectory for improving the current position estimate. Three localization schemes, which use the temporal information, are presented. The first is a recursive form of the Gauss method. The other two are extensions of the Kalman filter to the nonlinear problem at hand, namely, the extended Kalman filter and the unscented Kalman filter. These methods are compared with other algorithms, which do not make use of the temporal information. An extensive experimental study demonstrates the advantage of using the spatial-temporal methods. To gain some insight on the obtainable performance of the localization algorithm, an approximate analytical evaluation, verified by an experimental study, is conducted. This study shows that in common TDOA-based localization scenarios--where the microphone array has small interelement spread relative to the source position--the elevation and azimuth angles can be accurately estimated, whereas the Cartesian coordinates as well as the range are poorly estimated.

Robust Tracking of Multiple Sound Sources by Spatial Integration of Room And Robot Microphone Arrays

Abstract: Sound source tracking is an important function for a robot operating in a daily environment, because the robot should recognize where a sound event such as speech, music and other environmental sounds originates from. This paper addresses sound source tracking by integrating a room and a robot microphone array. The room microphone array consists of 64 microphones attached to the walls. It provides 2D (x-y) sound source localization based on a weighted delay-and-sum beamforming method. The robot microphone array consists of eight microphones installed on a robot head, and localizes multiple sound sources in azimuth. The localization results are integrated to track sound sources by using a particle filter for multiple sound sources. The experimental results show that particle filter based integration reduces localization errors and provides accurate and robust 2D sound source tracking.

Real-Time Tracking of Multiple Sound Sources by Integration of In-Room and Robot-Embedded Microphone Arrays

Abstract: Real-time and robust sound source tracking is an important function for a robot operating in a daily environment, because the robot should recognize where a sound event such as speech, music and other environmental sounds originate from. This paper addresses real-time sound source tracking by real-time integration of an in-room microphone array (IRMA) and a robot-embedded microphone array (REMA). The IRMA system consists of 64 ch microphones attached to the walls. It localizes multiple sound sources based on weighted delay-and-sum beam-forming on a 2D plane. The REMA system localizes multiple sound sources in azimuth using eight microphones attached to a robot's head on a rotational table. The localization results are integrated to track multiple sound sources by using a particle filter in real-time. The experimental results show that particle filter based integration improved accuracy and robustness in multiple sound source tracking even when the robot's head was in rotation

Learning a Precedence Effect-Like Weighting Function for the Generalized Cross-Correlation Framework

Abstract: Speech source localization in reverberant environments has proved difficult for automated microphone array systems. Because of its nonstationary nature, certain features observable in the reverberant speech signal, such as sudden increases in audio energy, provide cues to indicate time-frequency regions that are particularly useful for audio localization. We exploit these cues by learning a mapping from reverberated signal spectrograms to localization precision using ridge regression. Using the learned mappings in the generalized cross-correlation framework, we demonstrate improved localization performance. Additionally, the resulting mappings exhibit behavior consistent with the well-known precedence effect from psychoacoustic studies

Automatic audio gain control for concurrent capture applications

Abstract: A system level automatic gain control (“System AGC”) automatically initializes and controls analog microphone gain in an environment where multiple independent applications simultaneously receive an input from a single analog microphone or microphone array. In one embodiment, the System AGC also prevents those applications from acting to separately control the gain by intercepting external gain control commands and responding to the corresponding application with a corresponding digital gain applied to the input signal from the microphone. Consequently, the System AGC avoids problems relating to oscillations and instability in the microphone gain resulting from multiple applications trying to simultaneously control the gain while preventing each application from adversely affecting the quality of another application's audio capture signal. Further, in one embodiment, the System AGC also acts to maximize the signal to noise (SNR) ratio of the microphone without introducing clipping as a function of a sampled background environment.

Acoustic node calibration using a moving source

Abstract: Acoustic nodes, each containing an array of microphones, can track targets in x-y space from their received acoustic signals, if the node positions and orientations are known exactly. However, it is not always possible to deploy the nodes precisely, so a calibration phase is needed to estimate the position and the orientation of each node before doing any tracking or localization. An acoustic node can be calibrated from sources of opportunity such as beacons or a moving source. We derive and compare several calibration methods for the case where the node can hear a moving source whose position can be reported back to the node. Since calibration from a moving source is, in effect, the dual of a tracking problem; methods derived for acoustic target trackers are used to obtain robust and high resolution acoustic calibration processes. For example, two direction-of-arrival-based calibration methods can be formulated based on combining angle estimates, geometry, and the motion dynamics of the moving source. In addition, a maximum likelihood (ML) solution is presented using a narrowband acoustic observation model, along with a Newton-based search algorithm that speeds up the calculation the likelihood surface. The Cramer-Rao lower bound (CRLB) on the node position estimates is also derived to show that the effect of position errors for the moving source on the estimated node position is much less severe than the variance in angle estimates from the microphone array. The performance of the calibration algorithms is demonstrated on synthetic and field data.

A miniaturized adaptive microphone array under directional constraint utilizing aggregated microphones.

Abstract: This paper introduces a miniaturized microphone array using the Directionally Constrained Minimization of Power (DCMP) method, which utilizes the transfer functions of microphones located at the same place, namely aggregated microphones. The phased microphone array realizes a noise reduction and direction of arrival (DOA) estimation system according to differences in the arrival time, phase shift, and/or the level of the sound wave for each microphone. Hence it is difficult to miniaturize the microphone array. The objective of our research is to miniaturize the system size using aggregated microphones. In this paper, we first show that the phased microphone array system and the proposed aggregated microphone system can be described within the same framework. We then apply a microphone array under directional constraint to the aggregated microphones and compare the proposed method with the microphone array. We show the directional pattern of the aggregated microphones. We also show the experimental results regarding DOA estimation.

A method of accelerating the training of an acoustic echo canceller in a full-duplex beamforming-based audio conferencing system

Abstract: A method and apparatus is set forth for accelerating the total acoustic echo cancellation convergence time in a microphone array full-duplex conferencing system with beamformer. The invention is based on sequentially switching the beamformer to untrained sectors during periods of far-end speech activity and performing real-time cancellation of far-end echo signals from the near-end signals to consecutively adapt and store filter coefficients for the echo canceller corresponding to each sector.

Tracking talkers using virtual broadside scan and directed beams

Abstract: A communication system (e.g., a speakerphone) includes an array of microphones, a speaker, memory and a processor. The processor may be configured to perform acoustic echo cancellation, to track multiple talkers with highly directed beams, to design beams with nulls pointed at noise sources, to generate a 3D model of the physical environment, to compensate for the proximity effect, and to perform dereverberation of a talker's voice signal. The processor may also be configured to use a standard codec in non-standard ways. The processor may perform a virtual broadside scan on the microphone array, analyze the resulting amplitude envelope for acoustic source angles, examine each of the source angles with a directed beam, combine the beam outputs that show the characteristics of intelligence or speech.

Methods and apparatus for the targeted sound detection

Abstract: Targeted sound detection methods and apparatus are disclosed. A microphone array has two or more microphones M0 . . . MM. Each microphone is coupled to a plurality of filters. The filters are configured to filter input signals corresponding to sounds detected by the microphones thereby generating a filtered output. One or more sets of filter parameters for the plurality of filters are pre-calibrated to determine one or more corresponding pre-calibrated listening zones. Each set of filter parameters is selected to detect portions of the input signals corresponding to sounds originating within a given listening zone and filter out sounds originating outside the given listening zone. A particular pre-calibrated listening zone is selected at a runtime by applying to the plurality of filters a set of filter coefficients corresponding to the particular pre-calibrated listening zone. As a result, the microphone array may detect sounds originating within the particular listening sector and filter out sounds originating outside the particular listening zone.

A method to reduce training time of an acoustic echo canceller in a full-duplex beamforming-based audio conferencing system

Abstract: A method is set forth for reducing the total acoustic echo cancellation convergence time for all look directions in a microphone array based full-duplex system. The method is based on capturing the loudspeaker signal due to the first far-end speech bursts when the conferencing system is first used, as well as the corresponding loudspeaker feedback signals in the individual microphones. The captured signals are then used for consecutive adaptation of the acoustic echo canceller on all echo paths corresponding to all look directions of the beamformer, thereby training the AEC. This training process can be executed concurrently with normal phone operation, for example, as a background process that utilizes available processing cycles.

Effects of Angle of Attack and Velocity on Trailing Edge Noise Determined Using Microphone Array Measurements

Abstract: Trailing edge (TE) noise measurements for a NACA 63-215 airfoil model are presented, providing benchmark experimental data for a cambered airfoil. The effects of flow Mach number and angle of attack of the airfoil model with different TE bluntnesses are shown. Far-field noise spectra and directivity are obtained using a directional microphone array. Standard and diagonal removal beamforming techniques are evaluated employing tailored weighting functions for quantitatively accounting for the distributed line character of TE noise. Diagonal removal processing is used for the primary database as it successfully removes noise contaminants. Some TE noise predictions are reported to help interpret the data with respect to effects of flow speed, angle of attack, and TE bluntness on spectral shape and peak levels. Important findings include the validation of a TE noise directivity function for different airfoil angles of attack and the demonstration of the importance of the directivity function's convective ampli...

An Optimum Microphone Array Post-Filter for Speech Applications

Abstract: This paper proposes a post-filtering estimation scheme for multichannel noise reduction. The proposed method extends and improves the existing Zelinski’s and, the most general and prominent, McCowan’s post-filtering methods that use the auto- and crossspectral densities of the multichannel input signals to estimate the transfer function of the Wiener post-filter. A major drawback of these two speech enhancement algorithms is that the noise power spectrum at the beamformer’s output is over-estimated and therefore the derived filters are sub-optimal in the Wiener sense. The proposed method deals with this problem and can be considered as an optimal post-filter that is appropriate for a wide variety of different noise fields. In experiments over real-noise multichannel recordings, the proposed technique is shown to obtain a significant headstart over the other methods in terms of signal-to-noise ratio and speech degradation measures. In addition it is used for ASR experiments where promising preliminary results are presented. Index Terms: Speech enhancement, microphone array, post-filter, complex coherence, speech recognition.

Microphone array post-processor using instantaneous direction of arrival

Abstract: In this paper we describe a novel algorithm for postprocessing a microphone array’s beamformer output to achieve better spatial filtering under noise and reverberation. For each audio frame and frequency bin the algorithm estimates the spatial probability for sound source presence and applies a spatio-temporal filter towards the look-up direction. It is implemented as a real-time post-processor after a timeinvariant beamformer and it substantially improves the directivity of the microphone array. The algorithm is CPU efficient and adapts quickly when the listening direction changes. It was evaluated with a linear four element microphone array. The directivity index improvement is up to 8 dB, the suppression of a jammer 40° from the sound source is up to 17 dB.