Dual Microphone Research Articles

Sensor system for heart sound biomonitor

Heart sounds can be utilised more efficiently by medical doctors when they are displayed visually, rather than through a conventional stethoscope. A system whereby a digital stethoscope interfaces directly to a PC will be described along with signal processing algorithms adopted. The sensor is based on a noise cancellation microphone, with a 450Hz bandwidth and is sampled at 2250 samples per second with 12-bit resolution. Further to this, we discuss for comparison a piezo-based sensor with a 1kHz bandwidth. A major problem is that the recording of the heart sound into these devices is subject to unwanted background noise, which can override the heart sound and results in a poor visual representation. This noise originates from various sources such as skin contact with the stethoscope diaphragm, lung sounds (patient breathing), and other surrounding sounds such as speech. Furthermore we demonstrate a solution using ‘wavelet denoising.’ The wavelet transform is used because of the similarity between the shape of wavelets and the time-domain shape of a heartbeat sound. Thus coding of the waveform into the wavelet domain is achieved with relatively few wavelet coefficients, in contrast to the many Fourier components that would result from conventional decomposition. We show that the background noise can be dramatically reduced by a thresholding operation in the wavelet domain. The principle is that the background noise codes into many small broadband wavelet coefficients that can be removed without significant degradation of the signal of interest. Also explored is the use of a phase space representation of the heart sound to exploit different properties of the signal.

Use of microphone technology to improve user performance in noise.

Because of the improvement in SNR reported for dual microphones in the studies reviewed in this manuscript, it is felt that a greater number of manufacturers will be releasing producis containing dual-microphone technology in the near future. For example, at the time this manuscript was prepared, Siemens, ReSound and Danavox introduced dual-microphone BTE and ITE hearing aids with DSP. Micro-Tech also recently releascd an ITE with dual microphones. Oticon recently introduced an ITE with DSP and a D-Mic™. It is clear that more manufacturers will introduce hearing aids with dual microphones or the D-Mic™ in the years ahead.

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.]

Electrical compensation of the microphone sensitivities in a dual microphone directional hearing aid

Hearing aids with directional sensitivity have shown enhanced user satisfaction for their improved performance in noisy environments. One way to achieve the directional response patterns is to use two omnidirectional microphones with the necessary beamforming electronics to form a beam with the desired directionality. Proper operation of the beamforming circuitry requires that the frequency-dependent sensitivities of the two microphones be accurately matched, or corrected, in both magnitude and phase. This paper presents a method of electrically compensating the response of the two microphones to provide an accurate match for beamforming without the need for special sorting of the pairs for matched response. The circuit is suitable for field recalibration if aging causes the sensitivities to vary, and can also be implemented as an adaptive circuit to update the compensation parameters with time for any slow drift with time.

Noise canceling microphone with visual feedback

Briefly, according to the invention, a noise-cancelling microphone apparatus (400) is disclosed. The microphone (400) comprises a sound transducer (106) for converting sound waves to electrical signals and means for substantially preventing ambient noise from being converted to electrical signal. A circuit (100) is provided for determining the level of noise not prevented form being converted to electrical signal. The microphone (400) further includes a display (148) for indicating when the level of the ambient noise being converted to electrical signal is above a predetermined level.

Direction-sensing acoustic glass-break detecting system

Dual microphones and a time-of-arrival processing circuit comprise a direction-sensing system which improves the false alarm immunity of an acoustic glass break detector by restricting its coverage to a well-defined zone. By comparing the arrival times of an abrupt sound at two spaced microphones, the processing circuit determines the direction of the sound. If the sound originated in the intended coverage zone, the processing circuit generates a signal which enables the glass break detector; otherwise, the detector is inhibited. When the improved glass break detector is oriented so that potential false alarm sources are outside the coverage zone, false alarm immunity is enhanced.

Tubular bi‐directional microphone with flared entries

A bi-directional noise-cancelling microphone comprising a detector for detecting acoustical signals centrally positioned within a tubular member that has flared entries. The microphone provides a bi-polar response pattern, and effectively cancels unwanted noise incident upon the microphone at the sides of the tubular member, as well as noise above a particular frequency dictated by the length of the tubular member. By providing flared entries to the tube, the maximum peak response frequency of the microphone is extended up to 40% higher than that achieved with a similar tubular member with non-flared entries. An embodiment of the tubular bi-directional microphone with flared entries is disclosed wherein the microphone is provided with a two-stage pop filter system. The pop filter system includes a first screen covering the front entry of the microphone housing. A second screen is constructed so as to vent the first screen, thereby preventing pop signals from being produced.

User selectable noise canceling for portable microphones

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Adaptive enhancement of microphone array signals

The signal-to-noise ratio of a speech signal picked up by a microphone array can be improved by adaptive post processing. Enhancement techniques known from single microphone or dual microphone signal processing, like noise canceling and spectral subtraction can be extended to a multimicrophone array system. The noise canceling technique and derived structures try to model the room impulse response by an adaptive transversal filter. Thus the performance of these algorithms is limited by the ratio of filter length to reverberation time and by the capability to track the nonstationary impulse response. Reduction of the noise of approximately 8 dB can be achieved with acceptable filter length in a stationary environment, but precautions must be taken to avoid canceling of the desired speech signal. The spectral subtraction method yields higher improvements in signal-to-noise ratio up to 12 dB, but due to false estimation of magnitude short time spectra the processed speech signal contains distortion known as ‘‘musical tones.’’ Different structures for adaptive post processing of microphone array signals based on both enhancement techniques are presented and discussed. Their performance in stationary and nonstationary environments is characterized by means of improvement of the signal-to-noise ratio and subjective speech quality.

Study of phonetic classification/recognition performance on data collected with different microphones

The goal of this study is to seek an understanding of the effects of microphone variations on the MIT segment-based speech recognition system, summit. Specifically, phonetic classification and recognition performance are evaluated on utterances extracted from the timit corpus. The timit corpus offers phonetically-transcribed and time-aligned data for three different microphones—a Sennheiser close-talking, noise-canceling microphone, a Bruel and Kajar (B&K) far-field pressure microphone, and a telephone handset (plus channel distortion). These transducers cause different convolutional, additive, and bandwidth effects in the speech waveform. Experimental procedures are established to measure and analyze system performance under variable training and testing conditions. Classification uses Gaussian models on a feature vector consisting of Mel-frequency cepstral coefficients and their time derivatives, plus duration. The experiments show that performance in phonetic classification and recognition degrades from the Sennheiser (27% classification error) to the B&K (29%) and the telephone (43%). Performance further degrades when training and testing conditions are unmatched. Closer examination reveals that these degradations are concentrated in specific phonetic classes. For example, confusions between voiced and unvoiced fricative pairs account for a large percentage of the additional errors when training on the Sennheiser and testing on the B&K.

Workstation Robotics: A Pilot Study of a Desktop Vocational Assistant Robot

Rehabilitation robots are increasingly being viewed as an appropriate assistive technology interface for persons with disabilities. The Desktop Vocational Assistant Robot (DeVAR) system is a voice-controlled robotic workstation designed to enable persons with severe mobility impairments to function independently in a work environment. This study examined the overall efficacy of the DeVAR system, the level of expertise required for therapist and support personnel, routine maintenance requirements, and the readiness of the device for a multicenter evaluation. Two precommercial DeVAR workstations were installed at selected sites. The pilot study spanned 8 months in which staff members and four subjects with high-level quadriplegia evaluated the systems extensively. Responses were generally favorable regarding ease of use and overall training process. Subjects recommended modifications, including incorporation of a noise cancellation microphone, more effective use of workstation space, and addition of vocational tasks. All respondents thought that if the recommended changes were implemented, DeVAR would have potential as a vocational assistant. The small number of subjects was due to the limited subject pool available for the study. On the basis of pilot results, the Veterans Administration Rehabilitation Research and Development Service, Baltimore, Maryland, is conducting a national multicenter evaluation to determine the overall utility and commercial readiness of the DeVAR system.

A Multicenter Evaluation of the A&D TM-2420 Ambulatory Blood Pressure Recorde

The A&D TM-2420 (A&D Engineering, Milpitas, CA) is an automatic, portable, noninvasive blood pressure (BP) recorder which uses a dual microphone system for the detection of Korotkoff sounds. Its accuracy and clinical performance were assessed in a multicenter study that also addressed issues such as observer agreement and the effects of age, arm circumference, heart rate, posture, and blood pressure level on the observer-device differences. We compared 906 simultaneous, same-arm BP measurements in 151 subjects using the TM-2420 versus two skilled clinicians per site using a teaching stethoscope. The agreement between the TM-2420 and mercury column determinations were within 10 mm Hg for 86 to 91% of systolic readings and 91 to 94% of diastolic readings, depending on the posture; a level of agreement which would receive a 'B+' grade from the recent British Hypertension Society guidelines. The limits of agreement (2 standard deviations about the mean difference) for systolic BP between observers and the TM-2420 tended to be greater for the standing position (-20 to 15 mm Hg) compared to supine (-14 to 12 mm Hg) and seated (-13 to 8 mm Hg) positions. Limits of agreement between the observers and device were not dependent upon age, heart rate, arm size, or blood pressure level. Twenty-four-hour blood pressure monitoring in two of the four centers demonstrated an error code rate of 3.4%, excluding 'retries' that are one of the device's features. These data demonstrate an acceptable level of accuracy and performance of the sixth generation of the TM-2420 for use in clinical practice and research.(ABSTRACT TRUNCATED AT 250 WORDS)

The Effect of Various Hearing Protectors on Sound Localization in the Horizontal and Vertical Planes

Listeners were tested under free head/torso movement conditions, using a brief pulsed broadband noise signal presented from any 1 of 20 loudspeakers in two intersecting hemicircumferential arrays: one in the horizontal plane, one in the vertical. The listeners' task was to judge the whereabouts of each signal while listening in normal, open-ears conditions or while wearing various types of standard or electronic earmuffs or compressible foam earplugs. Results showed that listeners could accomplish the task successfully in normal (open-ear) conditions. Results using earplugs, nonelectronic earmuffs, and two types of dichotic (dual microphone) electronic earmuffs showed decrements in performance similar to each other. These decrements took the general form of loss of appreciation of the vertical plane and of front-rear differences in the horizontal plane. Lateral discrimination was blurred but otherwise intact. Of two diotic (single microphone) electronic earmuffs, one completely disrupted localization function and the other came close to it.

Noise-canceling microphone for video cameras

A novel noise-canceling microphone was developed for compact video cameras. Simple in structure, this microphone system enables elimination of acoustic noise generated by a noise source just behind the system, as well as vibration noise. It is confirmed that it is possible to achieve high-quality recorded sound with the system, and that it can be applied to other cases where reduction of noise generated by a source near the microphone is required. >

Acoustic noise analysis and speech enhancement techniques for mobile radio applications

This article examines speech enhancement techniques to be applied in a mobile radio environment. After a careful analysis of the noise characteristics inside a car, based on measurements of the actual acoustic field, some temporal and spatial noise suppression techniques are considered. Spectral magnitude noise subtraction has been cascaded to correlation-based cancellation; the advantages of this solution are discussed. The article compares the performance of a noise-cancelling microphone approach and signal processing methods.

Method for measuring noise-canceling microphone characteristics

A test method was developed using fast Fourier techniques to determine noise-canceling microphone nearfield frequency response and noise-canceling properties. This procedure utilizes transfer function analysis between a standard laboratory microphone and a noise canceling microphone. The standard microphone and the noise canceling microphone are mechanically coupled and move thru a pink noise sound field produced by an artificial voice. Transfer functions were measured at a point defined as the source of the sound field which yield the nearfield response and at fixed distances from that point which yield measures of noise canceling properties. A mapping of the sound field was completed by moving two standard laboratory microphones thru the sound field and measuring their transfer functions.

A Comparison of Mouth, Ear, and Contact Microphones

Four microphones, including an ear transducer and a bone oscillator, were used experimentally in simultaneous recording with a noise-cancelling microphone serving as a control. Speakers recorded PB word lists under three recording conditions of quiet, 100-db noise, and 111-db noise for each microphone comparison. Recordings were played back to a panel of 10 trained listeners over headphones under conditions of quiet and with white noise introduced into the circuit at an S/N ratio of 0 db. In quiet playback the control microphone maintained a fairly stable score within each set condition. For the quiet playback the control microphone was superior or equal to the other microphones for all recording conditions, with the ear-transducer and bone oscillator also showing superiority in the quiet recording conditions and one of the throat microphones demonstrating superior performance at 111-db recording condition. In the noise playback the control microphone did not perform consistently within each set condition. The somewhat tenuous results of this part of the experiment indicate that the ear transducer is superior at all recording levels for listening in noise with the throat microphone also demonstrating superior performance at 100-db and 111-db noise recording levels. Testing of variations of the original microphones substantiated the superiority of the noise-canceling microphone in noise recording conditions with playback in quiet. Fairly reliable performance for the ear transducer suggests its development for use in special situations. [This research was carried out in connection with Contract AF 19(604)-1577, monitored by Operational Applications Laboratory, Air Research and Development Command, Bolling Air Force Base, Washington 25, D. C.]