Pair Of Microphones Research Articles

Performance Improvement of TDOA-Based Speaker Localization in Joint Noisy and Reverberant Conditions

TDOA- (time difference of arrival-) based algorithms are common methods for speech source localization. The generalized cross correlation (GCC) method is the most important approach for estimating TDOA between microphone pairs. The performance of this method significantly degrades in the presence of noise and reverberation. This paper addresses the problem of 3D localization in joint noisy and reverberant conditions and a single-speaker scenario. We first propose a modification to make the GCC-PHAse transform (GCC-PHAT) method robust against environment noise. Then, we use an iterative technique that employs location estimation to improve TDOAs accuracy. Extensive experiments on both simulated and real (practical) data (in a single-source scenario) show the capability of the proposed methods to significantly improve TDOA accuracy and, consequently, source location estimates.

System And Method For Stereo Operation Of Microphones For Video Conferencing System

A microphone bar for obtaining audio input for a video conferencing system includes a master module and one or more slave modules. The master module houses a first pair of unidirectional microphones for obtaining audio from opposite sides of the bar, and each of the slave modules houses a second pair of unidirectional microphones for obtaining audio from opposite sides of the bar. Extensions connect the modules together, and at least one bus conducts audio signals through the extension from the microphones to the video conferencing system. The microphones are operable according to a broadside mode of stereo operation when a central axis of the bar is substantially orthogonal to a view angle of the conferencing system. Furthermore, the microphones are operable according to an end-fire mode of stereo operation when the central axis of the bar is substantially parallel to the view angle of the conferencing system.

2Pa2-4 Direction of Arrival Estimation Based on Delayed-sum Method Using a Pair of Microphone in Reverberation Environment(Poster Session)

2Pa2-4 Direction of Arrival Estimation Based on Delayed-sum Method Using a Pair of Microphone in Reverberation Environment(Poster Session)

A Modified SRP-PHAT Functional for Robust Real-Time Sound Source Localization With Scalable Spatial Sampling

The Steered Response Power - Phase Transform (SRP-PHAT) algorithm has been shown to be one of the most robust sound source localization approaches operating in noisy and reverberant environments. However, its practical implementation is usually based on a costly fine grid-search procedure, making the computational cost of the method a real issue. In this letter, we introduce an effective strategy that extends the conventional SRP-PHAT functional with the aim of considering the volume surrounding the discrete locations of the spatial grid. As a result, the modified functional performs a full exploration of the sampled space rather than computing the SRP at discrete spatial positions, increasing its robustness and allowing for a coarser spatial grid. To this end, the Generalized Cross-Correlation (GCC) function corresponding to each microphone pair must be properly accumulated according to the defined microphone setup. Experiments carried out under different acoustic conditions confirm the validity of the proposed approach.

A method of characterising performance of audio loudspeakers for linear alternator applications in low-cost thermoacoustic electricity generators

This paper investigates the feasibility of using commercially available loudspeakers as low-cost linear alternators for thermoacoustic applications, to convert acoustic power to electricity. The design of a purpose built experimental apparatus, in which a high intensity acoustic wave is induced by using a high power woofer, is described. The rig is used to excite loudspeakers (referred here as “alternators”) under test, while a pair of microphones and a laser displacement sensor are used to enable acoustic power measurements. The paper presents a case study in which characteristics of acoustic-to-electric energy conversion of a candidate loudspeaker (alternator) – selected from the viewpoint of general performance, as well as parameters such as: high force factor, low electrical resistance and low mechanical loss – are measured. The measurements of acoustic power absorbed by the alternator and the electric power extracted from it by the load resistor, which allow estimating acoustic-to-electric efficiencies, are presented. The alternator has been tested at different operating frequencies, cone displacements and load resistance values. The measurement results are discussed and compared in detail with the calculations based on the linear acoustics model.

Low-cost sound-based localization using programmable mixed-signal systems-on-chip

There have been extensive theoretical studies on sound-based localization using both, a pair of microphones and microphone arrays. In contrast, there has been much less work on implementing and experimenting sound-based localization realized as customized electronic designs. This paper presents a low-cost implementation of the sound-based localization method proposed in Halupka et al. [11]. A new method called wave counting is proposed in this paper, as an alternative to the Maximum Likelihood procedure used in [11]. The implementation uses PSoC programmable mixed-signal embedded system-on-chip, which incorporates microcontroller, on-chip SRAM and flash memory, programmable digital blocks, and programmable analog blocks, all integrated on the same chip. The paper presents an extensive set of experiments to characterize the quality of localization using the proposed low-cost design.

Aeroacoustic sound generation in simple expansion chambers

A method is presented for measuring the aeroacoustic source strength in ducts with flow at frequencies at which the wave motion can be considered substantially one-dimensional. The method is based on coherent power flux measurements using pairs of microphones positioned both upstream and downstream of the source region. The method is applied to a flow excited expansion chamber with aeroacoustic source measurements presented for chambers with a range of flow velocities and chamber lengths. The results indicate locked-on flow tones are generated in the chamber. The frequency of these locked-on flow tones is compared with that predicted using describing function theory applied to resonators with a grazing flow as well as that of other literature.

Localization of sound sources in rooms using an improved version of steered response power algorithm.

Localization of sound sources inside a room is a challenging problem. The possible applications involving speech‐based source localization systems range from teleconferencing to home automation systems. For example, the localization of a speaker inside a conference room can be very useful to place the speaker in a remote room by means of a spatial audio reproduction system. Also, new applications will appear in home automation if accurate source localization systems are available in the future. One of the most robust approaches to source localization is the SRP‐PHAT algorithm, which has shown to provide very good localization results inside rooms with moderate reverberation. However, the computational cost is highly dependent on the spatial sampling and the number of microphones, making very difficult the localization of sound events if a coarse spatial sampling is used. In this paper, we propose an improvement of this method where sound events are not missed even if a very coarse grid is used. The improvement is based on a previous calculation of the existent cross‐correlation lags between spatially adjacent points in the grid, assuring that the non‐sampled space is covered in terms of cross‐correlation lags between microphone pairs when running the algorithm. Several experiments conducted in different rooms with complex acoustics confirm the validity and benefits of the proposed method.

An Efficient, Time-of-Flight-Based Underwater Acoustic Ranging System for Small Robotic Fish

Small (decimeter-scale) robotic fish are promising mobile sensor platforms for aquatic environments. Fine-grained localization for dense networks of such robotic fish presents a challenge because of noisy underwater environment, required submeter accuracy, and constraints on onboard processing power and hardware complexity. In this paper, we present an efficient time-of-flight-based acoustic ranging system for localization of robotic fish with limited onboard resources. The system involves simple hardware: a single pair of monotone buzzer and microphone. The distance between two nodes is determined by the time it takes for an acoustic signal generated by the buzzer on the first node to reach the microphone on the second node. The arrival of the signal is detected with the sliding discrete Fourier transform (SDFT) algorithm, where the rise dynamics of the signal is modeled and used for compensation of detection latency. The algorithm is implemented onboard a small biomimetic robotic fish, and experiments in an indoor pool have shown that the compensated SDFT algorithm results in an underwater ranging error of 1.9 wavelengths (1 m), and is thus promising for localization of dense aquatic networks.

Broadband Source Localization From an Eigenanalysis Perspective

Broadband source localization has several applications ranging from automatic video camera steering to target signal tracking and enhancement through beamforming. Consequently, there has been a considerable amount of effort to develop reliable methods for accurate localization over the last few decades. Essentially, the localization process consists in finding the candidate source location that maximizes the synchrony between the properly time-shifted microphone outputs. In addition to using well known cross-correlation-based criteria such as the steered response power (SRP), minimum variance (MV), and multichannel cross-correlation (MCCC), this synchrony can also be measured using the averaged magnitude difference function (AMDF) and the averaged magnitude sum function (AMSF) whose calculations involve low computational cost. In earlier related works, the latter techniques have been used for time delay estimation (TDE) of a target source observed by only one pair of microphones. Their generalization to the multiple microphone case and application to source localization have not been studied yet. In this paper, we consider both categories, i.e., cross-correlation and AMDF (with AMSF)-based approaches, using an arbitrary number of microphones, and analyze their performance. Specifically, we first provide a unifying study of the most popular cross-correlation-based techniques, such as the SRP, MV, and MCCC. In this paper, we use the eigenanalysis of the parameterized spatial correlation matrix (PSCM) to classify these methods and gain some insight into their performance. We demonstrate, for instance, that the MV and SRP consist in searching the major eigenvalue of the PSCM, while the MCCC, essentially, combines its minor eigenvalues when scanning for the source location. Inspired by this analysis, we show, in the second part of this work, the efficiency of the AMDF and AMSF in localizing an acoustic source using multiple microphones. Indeed, we propose two new parameterized matrices named as the parameterized averaged magnitude difference matrix (PAMDM) and the parameterized averaged magnitude sum matrix (PAMSM). The eigenanalysis of these matrices also reveals new criteria for acoustic source localization. Simulation results are provided to illustrate the effectiveness of all the investigated and proposed methods.

Time domain wave separation using multiple microphones

Methods of measuring the acoustic behavior of tubular systems can be broadly characterized as steady state measurements, where the measured signals are analyzed in terms of infinite duration sinusoids, and reflectometry measurements which exploit causality to separate the forward and backward going waves in a duct. This paper sets out a multiple microphone reflectometry technique which performs wave separation by using time domain convolution to track the forward and backward going waves in a cylindrical source tube. The current work uses two calibration runs (one for forward going waves and one for backward going waves) to measure the time domain transfer functions for each pair of microphones. These time domain transfer functions encode the time delay, frequency dependent losses and microphone gain ratios for travel between microphones. This approach is applied to the measurement of wave separation, bore profile and input impedance. The work differs from existing frequency domain methods in that it combines the information of multiple microphones within a time domain algorithm, and differs from existing time domain methods in its inclusion of the effect of losses and gain ratios in intermicrophone transfer functions.

Active Noise Hybrid Time-Varying Control for Motorcycle Helmets

Recent noise at work regulations in the EU (2003) have been established to prevent noise induced hearing loss (NIHL). This imposes better performance results to traditional feedback active noise control (ANC) in motorcycle helmets, which suffer from well known limitations. Here two new ideas are applied to this problem. First, an hybrid (feedforward/feedback) linear time invariant (LTI) controller is designed for a motorcycle helmet ANC, which improves the resulting attenuation. This is achieved by adding an extra pair of microphones which measure the external noise that is then used as the feedforward input signal. In addition and to increase even more the resulting performance, the air velocity is measured in real-time and used as the parameter which schedules a linear parameter varying (LPV) feedback (FB) controller. This is combined with the previous feedforward (FF) controller, resulting in a time-varying hybrid controller. Both hybrid, LTI and LPV controllers are designed using linear matrix inequality (LMI)-based optimization. Two experiments have been carried out to measure the relation between external noise spectra and velocity: a wind tunnel test and a freeway ride experience. The resulting controllers are tested in a simulation which uses actual data obtained from the freeway experiment. The resulting attenuations in this motivating study seem promising for future controller tests to be performed in real-time, with the adequate hardware.

Voice-awareness control for a humanoid robot consistent with its body posture and movements

Abstract This paper presents voice-awareness control consistent with robot’s head movements. For a natural spoken communication between robots and humans, robots must behave and speak the way humans expect them to. The consistency between the robot’s voice quality and its body motion is one of the most especially striking factors in naturalness of robot speech. Our control is based on a new model of spectral envelope modification for vertical head motion, and left-right balance modulation for horizontal head motion. We assume that a pitch-axis rotation, or a vertical head motion, and a yaw-axis rotation, or a horizontal head motion, effect the voice quality independently. The spectral envelope modification model is constructed based on the analysis of human vocalizations. The left-right balance model is established by measuring impulse responses using a pair of microphones. Experimental results show that the voice-awareness is perceivable in a robot-to-robot dialogue when the robots stand up to 150 cm away. The dynamic change in the voice quality is also confirmed in the experiment.

A preliminary comparison of sound quality of maternal speech recordings in varied neonatal intensive care unit settings.

Since the 1980s, research investigating the beneficial effects of exposure to maternal voice has been steadily increasing. This has been accomplished by playing back recordings of maternal voice to the preterm infants. It is not known, however, how variations in the typical NICU setting alter the sound quality of the original recordings. This is a sub-study to a larger study in which 58 preterm infants will hear a recording of their mother reciting a rhyme twice a day from 28–34 weeks post-menstrual. A pair of microphones will be placed at ear level of the preterm subjects while the original recordings of maternal speech are played. The sound recorded within the incubator will then be run through a sound analyzer (PRAAT) and compared to the same sound qualities of the original recordings. The purpose of this study is to compare the original sound spectrum of a maternal speech recording to spectra obtained while the recording is played within typical NICU settings (open vs closed crib, pillows vs no pillows, etc.) If it is observed that the sound quality is altered due to varied NICU settings, this information will be important in determining NICU sound regulations for future studies employing voice recordings.

Sound Source DOA Estimation and Localization in Noisy Reverberant Environments Using Least-Squares Support Vector Machines

This paper presents two new algorithms for mapping the time-differences-of-arrival (TDOAs) measured from the microphone pairs to sound source direction-of-arrival (DOA) and location in room environments based on the least-squares support vector machine (LS-SVM). Least squares (LS) has been widely used in the TDOA based algorithms for sound source DOA estimation or localization to map the measured TDOAs into sound source DOA or location. The drawback of LS mapping is that its performance degrades significantly in some scenarios. To combat this problem, an LS-SVM regression based algorithm for the nonlinear mapping is proposed, which outperforms the LS based algorithm in noisy reverberant rooms. Conventional approaches to sound source localization usually assume that the microphones used are ideal and that the locations of the microphones are also known a priori, which may not be well satisfied in practice. Therefore, the microphone arrays need to be calibrated carefully before use. However, it is not an easy task to calibrate microphone arrays perfectly. In this paper, we also proposed an algorithm for sound source localization based on the LS-SVM, which has the advantage that microphone array calibration is not required. The performance of the proposed algorithms is validated by the simulation results in noisy reverberant environments.

Chatter prevention and improved finish of workpiece for a milling process

This paper presents how real-time chatter prevention can be realized by feedback of an acoustic cutting signal. The efficacy of the proposed adaptive spindle speed tuning algorithm is verified by intensive experimental simulations. A pair of microphones, perpendicular to each other, is used to acquire the acoustic cutting signal resulting from milling chatter. A real-time feedback control loop is constructed for spindle speed compensation in such a way to ensure that the milling process is within the stability zone of the stability lobe diagram. The acoustic chatter signal index (ACSI) and the spindle speed compensation strategy (SSCS) are proposed to quantify the acoustic signal and actively to tune the spindle speed respectively. By converting the acoustic feedback signal into the ACSI, an appropriate spindle speed compensation rate (SSCR) can be determined by the SSCS based on a real-time chatter level or the ACSI. Accordingly, the compensation command, referred to as added-on voltage (AOV), is applied to increase/decrease the spindle motor speed. By employing the commercial software MATLAB/Simulink and the dSpace DS1104 interface module to implement the controller, the proposed chatter prevention algorithm is practically verified by intensive experiments. By inspection on the precision and quality of the workpiece surface after milling, the efficacy of the real-time chatter prevention strategy via acoustic signal feedback is further assured.

Determining Mixing Parameters From Multispeaker Data Using Speech-Specific Information

In this paper, we propose an approach for processing multispeaker speech signals collected simultaneously using a pair of spatially separated microphones in a real room environment. Spatial separation of microphones results in a fixed time-delay of arrival of speech signals from a given speaker at the pair of microphones. These time-delays are estimated by exploiting the impulse-like characteristic of excitation during speech production. The differences in the time-delays for different speakers are used to determine the number of speakers from the mixed multispeaker speech signals. There is difference in the signal levels due to differences in the distances between the speaker and each of the microphones. The differences in the signal levels dictate the values of the mixing parameters. Knowledge of speech production, especially the excitation source characteristics, is used to derive an approximate weight function for locating the regions specific to a given speaker. The scatter plots of the weighted and delay-compensated mixed speech signals are used to estimate the mixing parameters. The proposed method is applied on the data collected in actual laboratory environment for an underdetermined case, where the number of speakers is more than the number of microphones. Enhancement of speech due to a speaker is also examined using the information of the time-delays and the mixing parameters, and is evaluated using objective measures proposed in the literature.

Measurement of Vertical Temperature Distribution Using a Single Pair of Loudspeaker and Microphone with Acoustic Reflection

It is important to maintain an adequate indoor temperature for comfortable working conditions, improvement of the rate of production of farm goods grown in greenhouses, and for saving energy. Thus, it is necessary to measure the temperature distribution to realize efficient air-conditioning systems. However, we have to use many conventional instruments to measure the temperature distribution. We proposed a measurement system for vertical temperature distribution using a single pair of loudspeaker (SP) and microphone (MIC), and acoustic reflectors. This system consists of SP, MIC, and multiple acoustic reflectors, and it can be used to determine the temperature distribution from the mean temperature of the area bounded by two reflectors. In experiments, the vertical temperature distribution was measured using five sound probes in a large facility every 20 s for 24 h. From the results of this experiment, it was verified that this system can be used to measure the vertical temperature distribution from the mean temperature of each area bounded by two reflectors. This system could be used to measure the change in the temperature distribution over time. We constructed a simple system to measure the vertical temperature distribution.

A transfer matrix method for estimating the dispersion and attenuation of plane waves in the standing wave tube.

In this paper, an iterative method based on the transfer matrix approach is described for evaluating the sound speed and attenuation constant of air in a standing wave tube. The procedure is based on transfer matrix, the four‐microphone method for measuring the transmission loss of acoustical materials. In the latter procedure, a sample is placed between two pairs of microphones, where the sample is exposed to an incident plane wave field. In the present case, the air inside the standing wave tube is treated as the sampled material. Since the knowledge of the dispersion characteristics of plane waves within the tube is required to perform this measurement, an iterative method must be applied at the postprocessing step to estimate the complex wave numbers in the sample section. The result from the experiments showed that the air temperature within the tube has a significant impact on the results. The measured results were found to be in good agreement with the Temkin formula for attenuation in tubes. In ad...

Spectral differences between live and replayed speech measured in occluded ear.

To further our understanding of the transfer function between the mouth and the eardrum for occluded ears, we compared the recordings of live and played-back self-generated speech captured in the speaker’s ear canals. The airborne speech feedback (during the speech production) and its recording (during the playback phase) were delivered binaurally via earmolds or insert earphones (ER-2, Etymotic Research, Inc.). Three earmold styles were tested: Full shell with the canal portion extending either to the first or the second bend of the participant’s ear canal, and a custom open-fit (Microsonic 19A). In four separate conditions, in-the-canal speech recordings were made using a pair of probe tube microphones (ER-7C, Etymotic Research, Inc.) that were inserted at a fixed distance from the eardrum. Two female speakers provided samples of four word-imbedded vowels (heed, head, hawed, who’d). The participants spoke at a comfortable speaking level, and their speech was presented at a sound pressure level calibrated to match the level at the ears. The equivalent sound level and long-term average spectra of the corresponding pairs of speech samples were then analyzed as a function of the ear occlusion type. [Support provided by the Rehabilitation Engineering Research Center (RERC) on Hearing Enhancement.]