Single Omnidirectional Microphone Research Articles

Aircraft noise monitoring using multiple passive data streams

Monitoring of aircraft in the skies surrounding an airport is important to ensure the safety of the aircraft as well as quality of life for nearby residents. The traditional method of tracking aircraft position involves the ubiquitous rotating active radar, and the traditional method of monitoring aircraft noise is the single omnidirectional microphone. Both of these systems have significant disadvantages. In the past several decades, microphone array acoustic monitoring systems have become widely known although they are typically practical only for research purposes. More recently, advanced beamforming algorithms have set the stage for much smaller compact microphone arrays. Meanwhile, the Automatic Dependent Surveillance-Broadcast position tracking system has quickly risen in popularity. In this study, an advanced compact microphone array is used along with Automatic Dependent Surveillance-Broadcast and more conventional methods to monitor and track aircraft that are approaching and departing Chicago’s Midway Airport. The systems are portable and compact, and the unified streams of data can more fully characterize an individual aircraft, offering distinct advantages over existing monitoring systems.

Uncertainty in room acoustics: A consultant's perspective

Faced with time and budget constraints on most projects, acoustical consultants make a series of choices when asked to evaluate the acoustics of an existing space. Measure the impulse response of the room with sophisticated technology (MLS, sine-sweeps, etc.), or pop a balloon? Use a dodecahedral loudspeaker (or several), a more standard portable amplified loudspeaker, or the house sound system (or pop a balloon)? Use a single omni-directional microphone, a binaural dummy head, or a B-format soundfield microphone? How many source locations, and how many receiver locations? Whither measure? There is a growing body of literature that identifies uncertainty and inconsistency in reverberation time and other room acoustics metrics and the tools used to obtain them. The authors will add modestly to this evidence from consulting experience, and then focus on how some consultants in architectural acoustics choose what data to gather, how to gather it, and to what extent. Considerations of time, budget, and logistics factor in those choices, as do determinations regarding the essence of the client's needs and concerns, subjective listening to determine what measurements could be most useful, and our own expectations for how we might use the measurements we make, now and in the future.

Parameter selection methods of delay and beamforming for cochlear implant speech enhancement

Speech acquisition by a cochlear implant (CI) is typically performed by a single omnidirectional microphone. In order to improve the signal-to-noise ratio (SNR) of sound collected from the forward direction, the delay beamforming method is applied to satisfy the size and processing limitations. The optimal delay time and weighting parameters, selected by our proposed graphical method, yielded a beam pattern that strongly enhanced speech from the front, while noise from lateral and rear directions were sharply weakened, making the method appropriate for use in CI applications.

A Two-Microphone Noise Reduction System for Cochlear Implant Users with Nearby Microphones—Part II: Performance Evaluation

Users of cochlear implants (auditory aids, which stimulate the auditory nerve electrically at the inner ear) often suffer from poor speech understanding in noise. We evaluate a small (intermicrophone distance 7 mm) and computationally inexpensive adaptive noise reduction system suitable for behind-the-ear cochlear implant speech processors. The system is evaluated in simulated and real, anechoic and reverberant environments. Results from simulations show improvements of 3.4 to 9.3 dB in signal to noise ratio for rooms with realistic reverberation and more than 18 dB under anechoic conditions. Speech understanding in noise is measured in 6 adult cochlear implant users in a reverberant room, showing average improvements of 7.9–9.6 dB, when compared to a single omnidirectional microphone or 1.3–5.6 dB, when compared to a simple directional two-microphone device. Subjective evaluation in a cafeteria at lunchtime shows a preference of the cochlear implant users for the evaluated device in terms of speech understanding and sound quality.

Bilateral cochlear implantation and directional multi-microphone systems

Better speech understanding in noise is a common goal of bilateral cochlear implantation and directional multi-microphone systems. In this study, the performance of five signal processing/presentation strategies is compared: (1) monaural presentation using a single omnidirectional microphone, (2) monaural presentation using a simple two-microphone beamformer in a behind-the-ear (BTE) housing, (3) monaural presentation using a complex four-microphone adaptive beamformer, (4) binaural presentation using two omnidirectional microphones and (5) binaural presentation using two simple two-microphone beamformers in separate behind-the-ear units. Speech reception thresholds (SRT) in noise were measured in four adults implanted bilaterally with MED-EL cochlear implant systems. Speech was presented from the front and speech spectrum noise from an azimuth of 90°. For all conditions involving microphones at only one ear, the noise source was placed at the contralateral side. Average speech reception threshold (SRT) improvements over the single omnidirectional microphone (condition 1) were 3.0 dB for the simple two-microphone BTE beamformer, 1.8 dB for bilateral cochlear implants and 3.6 dB for the combination of both. The largest mean improvement (13.0 dB) was achieved by the complex four-microphone beamformer. Bilateral cochlear implants and multi-microphone beamforming seem to complement each other but are outperformed by complex signal processing strategies in this laboratory setting.

Improved BTE hearing-aid directivity using a directional microphone array

Extraction of speech in noise is of great importance to hearing-impaired listeners. Directional microphones are incorporated in some hearing aids to improve noise rejection through increased directivity. A symmetric cardioid response is created either with a single directional microphone, or using beamforming with two omnidirectional microphones. The head-related transfer function (HRTF), however, introduces an asymmetry that cannot be exploited by a linear array of omnidirectional microphones. A new BTE array consisting of a gradient directional microphone with nulls in the front–back vertical plane and two omnidirectional microphones exploits the asymmetry of the HRTF to obtain almost 2 dB better directivity than the best cardioid. HRTFs measured on KEMAR with this array were transformed to the frequency domain, where directivity-maximizing coefficients in each band were derived. The Articulation-Index (AI) weighted directivity gain of this optimal three-element directional array was 6.4 dB greater than a single omnidirectional microphone on the BTE, whereas the directivity gain of the HRTF-optimized two-omni beamformer was 4.6 dB, and the optimal free-field cardioid placed on the head yielded 4.4 dB. [Work partially supported by NIH (NIDCD) under Grant No. 1 R01 DC005762-01A1.]

A directional adaptive least‐mean‐square acoustic array for hearing aid enhancement

This paper introduces directional microphones to adaptive array processing for hearing aid applications. Acoustic fixed arrays are designed to match a focused array gain pattern, while acoustic adaptive arrays are designed to attenuate interference noises with changing characteristics. However, as currently constructed, acoustic adaptive arrays cannot stay focused with a limited number of microphones available in a cosmetically acceptable hearing aid. In this paper, a technique is discussed that combines fixed and adaptive arrays in a system which enhances the desired signal while effectively attenuating interference speech and background noise. In particular, the design and performance of a directional adaptive least-mean-square (LMS) acoustic four-element array with a restricted geometry, where the array microphones are directional microphones, are examined. Simulations show that the directivity index of the directional adaptive array using four hypercardioid microphones is improved to between 8.6 and 11 dB. The array reduces the interference noises by 29.7 to 42.3 dB and provides a signal-to-noise ratio improvement of 11.5 to 12.2 dB over a single omnidirectional microphone. The sensitivity analysis is also discussed. It is concluded that the small size (four-element) microphone array can spatially filter interference noise effectively and so improve SNR performance significantly.

A directional adaptive LMS acoustic array for hearing enhancement

Acoustic fixed arrays are designed to match a focused array gain pattern, while acoustic adaptive arrays are designed to respond to changing environments. However, as currently constructed, adaptive arrays cannot stay focused with the limited number of microphones available in a cosmetically acceptable hearing aid. In this paper, a technique that combines fixed and adaptive arrays in a system that enhances the desired signal while effectively attenuating interference speech and background noises is discussed. In particular, the design and performance of a directional adaptive four-supercardioid microphone array is examined. This idea was formed independently from the use by Soede etal. [J. Acoust. Soc. Am. 94, 785–798 (1993)] of fixed arrays with cardioid microphones. Thus it is confirmed that using directional microphones improves array directivity in the hearing aid application. Simulations show that the directivity of this adaptive array is improved to 7.15–7.71 dB. The array reduces interferences from different directions by about 15–30 dB and provides a SNR improvement of about 9–12 dB over a single omni-directional microphone. A sensitivity analysis is also discussed. [Work partially supported by Deafness Research Foundation.]

Assessment of a directional microphone array for hearing-impaired listeners

Hearing-impaired listeners often have great difficulty understanding speech in surroundings with background noise or reverberation. Based on array techniques, two microphone prototypes (broadside and endfire) have been developed with strongly directional characteristics [Soede et al., "Development of a new directional hearing instrument based on array technology," J. Acoust. Soc. Am. 94, 785-798 (1993)]. Physical measurements show that the arrays attenuate reverberant sound by 6 dB (free-field) and can improve the signal-to-noise ratio by 7 dB in a diffuse noise field (measured with a KEMAR manikin). For the clinical assessment of these microphones an experimental setup was made in a sound-insulated listening room with one loudspeaker in front of the listener simulating the partner in a discussion and eight loudspeakers placed on the edges of a cube producing a diffuse background noise. The hearing-impaired subject wearing his own (familiar) hearing aid is placed in the center of the cube. The speech-reception threshold in noise for simple Dutch sentences was determined with a normal single omnidirectional microphone and with one of the microphone arrays. The results of monaural listening tests with hearing impaired subjects show that in comparison with an omnidirectional hearing-aid microphone the broadside and endfire microphone array gives a mean improvement of the speech reception threshold in noise of 7.0 dB (26 subjects) and 6.8 dB (27 subjects), respectively. Binaural listening with two endfire microphone arrays gives a binaural improvement which is comparable to the binaural improvement obtained by listening with two normal ears or two conventional hearing aids.

Ear‐adequate spectral analysis for noise evaluation

In numerous sound situations, sound quality cannot be determined by means of conventional measuring techniques. The frequent discrepancy between subjective judgment of sound events yielded by standard measurement procedures, can be explained by the difference between a sound situation as received by a single omnidirectional microphone, on the one hand, and as received by the human hearing, on the other hand. A new procedure using a dummy head and a special binaural analyzer has been developed for spectral analysis that realizes frequency and time resolution in the entire audio spectral range. This is comparable to the properties of human hearing, which is able to perceive quickly changing time structures and minor frequency differences at the same time. New approaches for ear‐adequate noise evaluation in order to judge noise quality can be achieved by means of psychoacoustic properties of human heating and binaural signal processing. The results of these investigations show that human hearing, which has two auditory paths with a complex directional characteristic and a binaural pattern recognition, gives a different judgment of noise quality as compared to conventional measurement methods, especially in complex sound situations with a spatial distribution of several sound sources and a sound‐pressure level below 85 dB.