Dual Microphone Array Research Articles

Speech Enhancement of Mobile Devices Based on the Integration of a Dual Microphone Array and a Background Noise Elimination Algorithm.

Mobile devices are often used in our daily lives for the purposes of speech and communication. The speech quality of mobile devices is always degraded due to the environmental noises surrounding mobile device users. Regretfully, an effective background noise reduction solution cannot easily be developed for this speech enhancement problem. Due to these depicted reasons, a methodology is systematically proposed to eliminate the effects of background noises for the speech communication of mobile devices. This methodology integrates a dual microphone array with a background noise elimination algorithm. The proposed background noise elimination algorithm includes a whitening process, a speech modelling method and an H2 estimator. Due to the adoption of the dual microphone array, a low-cost design can be obtained for the speech enhancement of mobile devices. Practical tests have proven that this proposed method is immune to random background noises, and noiseless speech can be obtained after executing this denoise process.

Background Noise Reduction Design for Dual Microphone Cellular Phones: Robust Approach

Cellular phones are used almost everywhere for the communication purpose around the world. However, until now, in noisy environments, the speech communication quality of cellular phones is still awful because the noisy environments where people truly talk are quite complicated and there exists no suitable background noise reduction solution for dealing with this problem effectively. Based on these reasons, this research successfully proposes a novel noise reduction methodology to solve the background noise corrupted problem of cellular phones via a systematic design that comprises a dual microphone array hardware and a novel algorithm combined with a background noise whitening process, the system modeling method, and the robust estimation concept software. After practical experiments, a fact that this proposed method has the immunity to the really noisy environment where cellular phones are used can be found. This background noise reduction design possesses the following impressive advantages: an average of a 20 dB enhanced noise reduction performance the averaged value of state-of-the-art methods are about 10-15 dB, and an average similarity of 93% of pure speeches can be obtained for the purified speeches by applying this proposed method. Besides, this proposed method can be easily implemented in any of the one shelf or newly developed cellular phones with a really low cost.

Real-time monitoring of welding path in pulse metal-inert gas robotic welding using a dual-microphone array

Online monitoring of welding path is important as deviation from the welding path will significantly affect the welding quality. This paper proposed a welding path monitoring system using a dual-microphone array for the pulse metal-inert gas (MIG) robotic welding process. A calibration method was developed for microphones to find best locations for collectin arc sound signals. Arc sound features were analyzed to obtain the relationship between the arc sound signal and the welding path deviation. Prediction models for welding paths were then established in two ways: (1) the prediction model was based on individual microphone signals and (2) the model was based on signals from the dual-microphone array which proved to have better prediction accuracy in monitoring the welding path. The dual-microphone model showed its potential advantage in quality monitoring and control for the MIG welding process. The prediction error could reach 0.6708 and 1.1901 mm. The monitoring system based on a dual-microphone array could meet the position accuracy requirement of welding quality control in pulse MIG welding.

TOA-Based Self-Calibration of Dual-Microphone Array

In this paper, we study the time-of-arrival (TOA) based self-calibration problem of dual-microphone array for known and unknown rack distance, and also for different combinations of dimension for the affine spaces spanned by the receivers and by the senders. Particularly, we analyze the minimum cases and present minimum solvers for the case of microphones and speakers in 3-D/3-D, in 2-D/3-D, and in 3-D/2-D, with given or unknown rack length. We identify for each of these minimal problems the number of solutions in general and develop efficient and numerically stable, non-iterative solvers. Solving these problems are of both theoretical and practical interest. This includes understanding what the minimal problems are and how and when they can be solved. The solvers can be used to initialize local optimization algorithms for finding the maximum likelihood estimate of the parameters. The solvers can also be used for robust estimation of the parameters in the presence of outliers, using, e.g., RANSAC algorithms. We demonstrate that the proposed solvers are numerically stable in synthetic experiments. We also demonstrate how the solvers can be used with the RANSAC paradigm. We also apply our method for several real data experiments, using ultra-wide-band measurements and using acoustic data.

A reverberation robust target speech detection method using dual-microphone in distant-talking scene

Speech signal processing with coherent interference in reverberant environment under distant-talking scene has always been a difficult problem, in which Target Speech Detection (TSD) plays a basic role. This paper proposes a reverberation robust TSD method, which is based on Beam-to-Reference Ratio (BRR) using a dual-microphone array, is proposed. At first, detection thresholds in Time–Frequency (T–F) domain are derived under free sound field assumption. A novel estimator, Direct-to-Reverberate Ratio (DRR), is introduced to enlarge the basic assumption to reverberant environment which is common in distant-talking scene. Then T–F thresholds of BRR are revised according to DRR. Meanwhile, innate weak point of compact array is studied due to spatial aliasing and a sidelobe suppression procedure is proposed to further eliminate the effect of the coherent interference. According to these techniques, a state-of-the-art full-band judgement is obtained by statistics of likelihood on each T–F block. Experimental results show that the proposed method performs robust in different reverberant environments with coherent interferences when target speech is from the a priori known direction-of-arrivals (DOA) in distant-talking scene.

Panel Contribution Analysis in a Vehicle Cabin using a Dual Layer Handheld Array with Integrated Position Measurement

The paper describes an array-based method for measurement of panel contributions in a cabin. Both the FRF measurements and the operational measurements are taken with a dual layer microphone array, and the required sound field data on a panel surface mesh are calculated using the SONAH patch holography algorithm. Multiple array positions can be measured to cover the needed mesh area, and area-integration of contributions is then performed on the surface mesh. Measurement or alignment of surface mesh geometry and measurement of array positions is performed using an integrated 3D position measurement system. After a theoretical description the method is evaluated through simulated and actual measurements.

The Study on Comparison of Deduction Effects for Rear Background Noise between Dual and Triple Microphone Array

The Study on Comparison of Deduction Effects for Rear Background Noise between Dual and Triple Microphone Array

Turbulence Structures and the Acoustic Far Field of a Mach 1.3 Jet

The temporal characteristics of the acoustic far field of a Mach 1.3, high-Reynolds-number, ideally expanded axisymmetric jet and their potential correlation with large-scale turbulence structures within the jet were explored. A dual microphone array, placed approximately 30 deg from the jet axis in the acoustic far field, was used to determine the temporal variations of the acoustic field and the approximate locations of intense noise sources within the jet, as well as the time of noise emission with respect to the acquired planar flow images. Simultaneous double-pulse flow visualizations were used to identify turbulence structures, as well as their development and interaction in the region of intense noise generation. The time history of the acoustic data showed individual large-amplitude noise events, periodic large-amplitude noise events, and long periods of relative quiet without any large-amplitude noise production

Clinical trials of a hybrid adaptive beamformer (HAB) for improved speech understanding in noise

A wearable, hybrid adaptive beamformer (HAB) device has been developed using a four-microphone array and a combination of adaptive and fixed-weight beamforming. The HAB is being evaluated with hard-of-hearing (HoH) listeners in a variety of environments. Pilot tests used HINT test sentences and noise from separate loudspeakers (106-deg angle) in an audiological booth (AI-weighted direct-to-reverberant ratio 9.1 dB). Measured sentence reception thresholds (SRTs) indicated that the HAB provided 14.5, 16.1, and 12.8 dB improvement over the single microphone for two normal-hearing and one HoH listener. Results will be reported for ongoing objective and subjective testing of elderly listeners with mild to moderate hearing losses. Subjects are fitted monaurally with a commercially available behind-the-ear hearing aid with a dual microphone array following the NAL-R fitting algorithm. Comparisons are made between listeners’ SRTs in noise, using: (a) conventional single-microphone, (b) dual microphone, and (c) four-microphone HAB devices. HINT sentences are presented with single and dual noise sources at two arrival angles in three room environments: (a) sound booth, (b) favorable environment, and (c) noisy/reverberant environment. Listeners rate the three hearing aid systems and indicate their preference in paired comparison testing. SRTs, ratings, and preference judgments will be reported. [Work supported by NIH.]