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Background Although hearing aids (HAs) and cochlear implants (CIs) can provide significant benefits to persons with hearing loss, users frequently report difficulty hearing in noisy environments, particularly when there are multiple talkers. Little is known about the benefits provided by currently available wireless microphones in multitalker situations. Purpose The purpose of this study was to compare the benefits received in speech recognition in noise by adults with hearing loss when using two different wireless microphone types in a simulated group setting. Research Design A quasi-experimental, repeated-measures design was used where performance in a control condition, HA/CI alone, was compared with performance in two wireless microphone intervention conditions. Study Sample Participants included ten listeners, aged 20-92 years, with bilateral sensorineural hearing loss who were experienced HA or CI users. Intervention The two wireless microphones by Phonak, Roger Pen, and Roger Select used the same digital modulation protocol to transmit the signal to compatible receivers. However, the Roger Pen operated in a fixed omnidirectional mode, whereas the Roger Select operated in an adaptive directional mode. Data Collection and Analysis Participants were asked to repeat Hearing in Noise Test sentences presented in restaurant noise in three conditions: HA/CI alone, HA/CI with a Roger Pen, or HA/CI with a Roger Select microphone placed in the center of a round table. Sentences were presented from one of five loudspeakers equally spaced with the participant, while restaurant noise was presented on each side at four signal-to-noise ratios (SNRs), including +5, 0, −5, and −10 dB. A two-way, repeated-measures analysis of variance was performed with main effects of listening condition and noise level. Results Significantly greater speech recognition performance was achieved with the wireless microphones than with listening with just the HA or CI. Furthermore, at the −5- and −10-dB SNR conditions, the Roger Select resulted in significantly better performance than the Roger Pen microphone. Conclusions The results suggest that the Roger Select microphone can provide significant benefits in speech recognition in noise over the use of HA/CI alone (61%) and also significant benefits over the use of a Roger Pen (16%) in a simulated group dining experience.

Simple and professional audio mixers can be built using a collection of software and hardware technology components. In this article, software-defined radio (SDR) is used to build a wireless microphone audio-mixing system in combination with a market-available audio console. The RTL-SDR is inexpensive and used to receive signals in the radio-frequency (RF) spectrum. Thus, as a professional solution, this article develops a standalone audio mixer either for wireless microphones or for working cooperatively with audio consoles. The universal serial bus (USB)-based SDR hardware component employs an SDR to perform personal computer (PC)-based computations, and the free open-source software (OSS) GNU radio companion (GRC) is chosen to build the block-diagram of the SDR application. Another combination employs the extensible messaging and presence protocol (XMPP) for voice-over Internet protocol (VoIP) audio chatting. Therefore, the following OSSs are employed: Jitsi for the XMPP client in a PC, Empathy for another client in a laptop, and Openfire for the XMPP server. The three technologies and the combination of hardware and software components were successfully tested in different scenarios and proved that this solution can provide inexpensive and various options for developers, audio technicians, musicians, and hobbyists.

Objectives Although cochlear implantation has significantly contributed to the speech perception of cochlear implant (CI) users, these individuals still have significant difficulty in understanding speech, especially in noisy environments and keeping track of the target speaker in the presence of speech sounds of others. This study was aimed to evaluate the effect of wireless Remote Microphones (RM) on speech discrimination scores in noise in child CI users. Materials and methods Twenty children with unilateral cochlear implantation were enrolled in this study with mean ± SD age of 5.8 ± 0.83 years who have undergone CI for at least one year. Speech discrimination scores in noise were assessed using the Words-in-Noise (WIN) test at a constant signal-to-noise ratio (SNR) of 0 dB, in the presence and absence of a wireless RM. Three loudspeakers were placed at a distance of 1 m in front of the child to present the speech and babble noise. The wireless microphone was placed on a base with a height equivalent to the height of the middle speech loudspeaker at a distance of 30 cm from it. Findings The mean speech discrimination score in noise in the absence of wireless RM in all children was obtained 34% (6.8 words out of 20 words), with minimum and maximum of 15% and 50% words. Findings revealed the mean speech discrimination score in noise in the presence of wireless RMs is equivalent to 65% (13 words out of 20 words), with the minimum and maximum scores of 35% and 95%, respectively. The result showed that speech discrimination scores in noise improved in the presence of wireless RM. Conclusion The significant improvement was observed in speech discrimination in noise in all cochlear implanted children when the wireless RM was used, as compared to the absence of a wireless RM, which suggests the usefulness of this hearing aid accessory in CI users.

Generation Z uses technology frequently and demands learning that is individualized, engaging, digital, and visually based. They prefer learning videos to learn about the product making process. The research objectives were to develop the video of making mango sorbet for the Agriculture Product Processing course and evaluate the feasibility of the video. This was a research & development (R&D) study using 4D models (Define, Design, Develop, and Disseminate). The video was captured using Sony NX100 camera and Kino Flo 6-bank lighting. The presenter and voice-over dubber’s sound were recorded using Boya BY WM8 wireless microphone. The video was edited by Adobe Premiere Pro CC 2017. The video had a duration of 14 minutes 29 seconds with mp4 format. The video consisted of the opening section, the content section, and the closing section. The opening and closing section were delivered by the presenter. The content section was demonstrated by an actress and voice-over dubber. The video feasibility was evaluated by 2 material experts, a media expert, and 30 students of Vocational High School (VHS) 1 Cangkringan. The data were analysed descriptively. According to the feasibility evaluation, the making mango sorbet video was considered to be very feasible to be utilized as a learning media.

Spectrum sensing is one of the major challenges for commercial development of cognitive radio systems, since the detection of the presence of a primary user is a complex task that requires high reliability. This work proposes a signal classifier capable of detecting and identifying a primary user signal on a given channel of the radio spectrum. The proposed approach combines eigen-decomposition techniques and neural networks not only to decide about the presence of a primary user, but also to identify the primary user signal type, a feature that is not encountered in the current approaches proposed in literature. Besides the advantage of identifying the primary user type, the proposed method also considerably reduces the computational cost of the detection process. The proposed classification method has been applied to the development of five primary user signal Classification Modules, which includes wireless microphone, orthogonal frequency-division multiplexing and Digital Video Broadcasting-Terrestrial signals. The results show that the proposed classifier correctly detects and identifies the primary users, even under low signal to noise ratio and multipath scenarios.

1.To understand both proximate and ultimate factors shaping vocal communication, it is fundamental to obtain reliable information of participating individuals on different levels: Firstly, it is necessary to separate and assign the individuals’ vocalisations. Secondly, the precise timing of vocal events needs to be retained. Thirdly, vocal behaviour should be recorded from undisturbed animals in meaningful settings. A growing number of studies used animal-attached microphones to tackle these issues, but the implications for the study species and the research question often receded into the background. Here we aim to initiate a discussion about the limitations, possible applications and the broader potential of such methods. 2.Using lightweight wireless microphone backpacks (0.75 g including customised leg-loop harness) combined with multi-channel recording equipment we captured vocal behaviour of small songbirds. We evaluated the effect of the devices at various levels, including an assessment of how vocal and locomotor activities were affected by initial device attachment and battery exchange. We compared our approach to existing studies and identified suitable research examples. 3.We acquired continuous vocalisation recordings of zebra finches, and unequivocally assigned them to interacting individuals, with system-based synchrony, irrespective of background noise. We found effects of initial backpack attachment and of battery replacement on vocal and locomotor activity, but they were minimised through the extended recording duration (ca. 16 days) that outlasted habituation effects (ca. 3 days). 4.This method provides the tools to integrate individual vocal communications into a group setting, while enabling animals to behave freely in undisturbed, structured and acoustically complex environments. By minimising the effects on the animals, the behaviour under study, and ultimately on the research question, this approach will revolutionize the ability to capture individual-level vocalisations in a variety of communication contexts, opening up many new opportunities to address novel research questions. This article is protected by copyright. All rights reserved.

In this paper, we propose a low-complexity multistage polyphase filter bank for the detection and estimation of center frequency of wireless microphone (WM) in television channels for cognitive radios. The detection precision is directly related to the number of subbands (granularity) in the filter bank. This implies that the computational complexity becomes high if the number of stages and granularity of the filter bank structure are increased. The novelty of the proposed method is the estimation of center frequency with higher precision and reduced computational complexity using the centroid method. The merit of the centroid method is that the presence of WM can be detected along with estimation of center frequency in the first stage, if the WM lies partly in one subband and partly in the adjacent subband. This single-stage detection and estimation of WM significantly reduces computational and hardware complexities as well as latency. However, if the WM appears anywhere exclusively within a single subband, the detection process can be completed in the second stage without ambiguity. A mathematical expression for calculating the center frequency of WM from the subband energy (power) using centroid method is also derived and presented. The proposed scheme is analyzed and validated through extensive simulations for the detection of WM. The error in estimation of center frequency in most of the cases is less than 4 % for SNR variations from 0 to ź20 dB.

Background: Speech understanding in noise is challenging for individuals with hearing loss. Hearing aids (HAs) alone are typically unable to resolve these listening difficulties. Frequency modulation (FM) systems or other remote microphone accessories, coupled to HA, are intended to provide listeners with a good signal-to-noise ratio (SNR), thus improving signal audibility and speech understanding. Purpose: The goal of this study was to assess variables that influence SNR at the tympanic membrane (TM) when using a remote microphone/HA combination. We examined microphone setting, transmission system gain, and background noise levels using (1) mathematical computations to manipulate variables and observe the outcomes and (2) behavioral testing. Research Design: This study used mathematical computations to estimate SNR at the TM and a mixed-model experimental design to confirm a subset of the calculations. Study Sample: Ten children with normal hearing (mean age, 13.7 yr) and ten adults with high-frequency sensorineural hearing loss (mean age, 49.6 yr) participated. Data Collection and Analysis: Speech recognition thresholds were obtained using Bamford–Kowal–Bench sentences in the presence of noise. Participants used an FM system coupled to an HA in an FM-only and an FM + HA microphone condition. Results: Better performance was observed in the FM-only compared to FM + HA condition with the overall amount of the FM-only advantage slightly larger than the mathematical calculations predicted. Further calculations demonstrated that (1) when using an FM-only microphone setting, the SNR at the TM is determined primarily by the SNR at the FM microphone; (2) when both HA and FM microphones are active, the SNR is determined by the highest level of the speech, which is typically at the FM microphone, and the highest level of noise at either the FM or HA microphone; (3) increasing FM gain has no impact on SNR in an FM-only condition; and (4) in an FM + HA condition, increasing FM gain improves SNR. The amount of improvement depends on noise levels at the FM and HA microphones. When the noise levels are similar at the two microphones, an improvement in SNR of ˜2 dB is expected. Greater improvement is expected when the level of the noise at the FM microphone can be reduced relative to the level at the HA microphone. Conclusions: When using a remote microphone system coupled with a listener’s HA, several variables influence SNR at the TM. Two variables that can be manipulated by programming of either or both devices are the microphone setting and gain setting. Mathematical calculations were used to determine the specific influence of and interactions between these variables and showed the importance of (1) managing noise levels to optimize SNR; and (2) counseling clients regarding optimal use of and realistic expectations from their system. This information is useful in the clinical management of persons with hearing loss, especially with the advent and affordability of wireless microphone accessories to assist listeners in background noise.

Background Previous research showed benefits of remote wireless technology in bilaterally moderate- to-severe hearing-impaired participants provided with hearing aid(s), cochlear implant(s) (CIs), or bimodal devices as well as in single-sided deaf (SSD) cochlear implant recipients (with CI from Cochlear™) and normal-hearing (NH) participants. Purpose To evaluate the effect of the digital remote wireless microphone system, Roger™, on speech recognition at different levels of multisource noise in SSD CI recipients using MED-EL CI sound processor OPUS 2. Outcomes were assessed as a function of the listening condition (NH only, NH + CI, NH + CIRog, NHRog + CI, and NHRog + CIRog), Roger™ receiver type (Roger™ Focus for NH; Roger™ Xand Roger™ MyLink for CI) and accessory mixing ratio. Study Sample Eleven adult, SSD participants aided with CI from MED-EL. Data Collection and Analysis Speech recognition in noise was assessed in two no-Roger™ conditions, one Roger™ X condition, and two Roger™ MyLink conditions. For the Roger™ X and no-Roger™ conditions, speech recognition was tested at 60.3 dB(A) with the Oldenburg Sentence Test in classroom noise at levels of 55, 65, and 75 dB(A). For the two Roger™ MyLink conditions, speech recognition at 60.3 dB(A) was measured at a noise level of 75 dB(A). Roger™ X was assessed with an accessory mixing ratio of 1:1 (summation of unattenuated microphone and audio accessory input). For Roger™ MyLink, two accessory mixing ratios were investigated, MT (1:1, summation of unattenuated microphone and telecoil input) and T with maximum attenuation of microphone input. Results Speech recognition at higher noise levels (65 and 75 dB(A)) improved significantly with Roger™ in both unilateral use conditions (NH + CIRog and NHRog + CI) as well as bilateral use condition (NHRog + CIRog). Both the bilateral application of Roger™ and the unilateral Roger™ application on the NH ear outperformed the Roger™ application on CI alone. There was no statistically significant effect of type of CI Roger™ receiver (Roger™ X or Roger™ MyLink) and the accessory mixing ratio (MT or T) on speech recognition. Conclusions Speech recognition for distant speakers in multisource noise improved significantly with the application of Roger™ in SSD CI recipients. Both the unilateral Roger™ application on the NH ear or the CI as well as the bilateral Roger™ application can be recommended.

Throwable wireless microphone is an emerging technology to adopt in a teaching environment. It aims to promote classroom interaction. In this study, we examined students' learning experience and their perceptions regarding participation and interaction when adopted the technology in a large class setting at higher education level. Quasi experimental approach was employed to determine the impact of the technology on students. A total of 187 graduate students from a research oriented university in Hong Kong participated in the experiment. They were divided into control and subject groups. Instructor facilitated discussion questions and invited approximately half of the class to response via random selection in control group and throwing the microphone to random student in subject group. Quantitative feedbacks were gathered through two questionnaires to study their perceptions on learning experience and classroom interaction. Statistical analysis was conducted on the data and it revealed that the subject group, who was exposed to the technology, reported better learning experience mean score as compared to the control group. It was also discovered that active students in the subject group, who had actually caught the throwable microphone and talked in class, perceived significantly better classroom interaction compared to those merely sat and observed in the same classroom. The findings of this study suggested that the use of throwable wireless microphone technology indeed enhanced learning experience and classroom interaction in a large class environment. However, these positive impacts were not evenly effecting all participants. Participants who had caught-and-talked in class benefited the most from the adoption of the technology. In order to maximize the effect of the throwable microphone technology, instructors should increase the number of caught-and-talked participants in a large class setting.

An ANN based spectrum sensing technique for audio FM and the wireless microphone signals in TV band is proposed. The artificial neural network (ANN) model trained with the autocorrelation peaks of the signal in channel identifies it as a white space or a primary signal. The performance of this technique is efficient in terms of false alarm rate and probability of detection; the proposed method presents less mathematical complexity as compared to other recent spectrum sensing techniques. Simulation results are presented.

To evaluate whether speech recognition in noise differs according to whether a wireless remote microphone is connected to just the cochlear implant (CI) or to both the CI and to the hearing aid (HA) in bimodal CI users. The second aim was to evaluate the additional benefit of the directional microphone mode compared with the omnidirectional microphone mode of the wireless microphone. This prospective study measured Speech Recognition Thresholds (SRT) in babble noise in a 'within-subjects repeated measures design' for different listening conditions. Eighteen postlingually deafened adult bimodal CI users. No difference in speech recognition in noise in the bimodal listening condition was found between the wireless microphone connected to the CI only and to both the CI and the HA. An improvement of 4.1 dB was found for switching from the omnidirectional microphone mode to the directional mode in the CI only condition. The use of a wireless microphone improved speech recognition in noise for bimodal CI users. The use of the directional microphone mode led to a substantial additional improvement of speech perception in noise for situations with one target signal.

In this paper, we consider the problem of estimating the target sound direction of arrival (DoA) for a hearing aid (HA) system, which can connect to a wireless microphone worn by the talker of interest. The wireless microphone 'informs' the HA system about the noise-free target speech. To estimate the DoA, we consider a maximum-likelihood approach, and we assume that a database of DoA-dependent relative transfer functions (RTFs) has been measured in advance and is available. The proposed DoA estimator is able to take the available noise-free target speech, ambient noise characteristics, and the shadowing effect of the user's head on the received signals into account, and it supports both monaural and binaural microphone array configurations. Moreover, we analytically analyze the bias in the proposed estimator and introduce a modified estimator, which has been compensated for the bias. We demonstrate that the proposed method has lower computational complexity and better performance than recent RTF-based estimators. Furthermore, to decrease the number of parameters required to be wirelessly exchanged between the HAs in binaural configurations, we propose an information fusion strategy, which avoids transmitting microphone signals between the HAs. An important benefit of the proposed IF strategy is that the number of parameters to be exchanged between the HAs is independent of the number of HA microphones. Finally, we investigate the performance of variants of the proposed estimator extensively in different noisy and reverberant situations.

This Paper presents the results of spectral observations in the UHF TV Bands IV and V from 470 MHz up to 862 MHz with focus on the TV-Channels 61 to 63 and 67 to 69. Concerning the discussions on \\citet{WRC} this frequency range is in great demand of several applications and is usually treated as a "white space" in the TV-Bands. According to typical scenarios, two different spectral loads will be presented considering the requirements of professional wireless microphone receivers with respect to in-band intermodulation.

Little is known about the perception of artificial spatial hearing by hearing-impaired subjects. The purpose of this study was to investigate how listeners with hearing disorders perceived the effect of a spatialization feature designed for wireless microphone systems. Forty listeners took part in the experiments. They were arranged in four groups: normal-hearing, moderate, severe, and profound hearing loss. Their performance in terms of speech understanding and speaker localization was assessed with diotic and binaural stimuli. The results of the speech intelligibility experiment revealed that the subjects presenting a moderate or severe hearing impairment better understood speech with the spatialization feature. Thus, it was demonstrated that the conventional diotic binaural summation operated by current wireless systems can be transformed to reproduce the spatial cues required to localize the speaker, without any loss of intelligibility. The speaker localization experiment showed that a majority of the hearing-impaired listeners had similar performance with natural and artificial spatial hearing, contrary to the normal-hearing listeners. This suggests that certain subjects with hearing impairment preserve their localization abilities with approximated generic head-related transfer functions in the frontal horizontal plane.

Most wireless microphones operate on vacant television broadcasting channels in the very high frequency and ultrahigh frequency bands, as secondary users operating on the no-interference and no-protection basis. Some wireless microphones share the 2.4 GHz band with other wireless devices. The 698–806 MHz band was repurposed when television broadcasting switched from analog to digital (and the 600 MHz band may be repurposed in the near future), resulting in less spectrum available for wireless microphone operations. To keep wireless microphones operating effective in this environment, engineers, researchers, and regulators require comprehensive knowledge of the spectrum, interference, and regulations regarding wireless microphones. This paper first presents the characteristics of the frequency bands currently allocated for wireless microphone operations. Next, interference that may be experienced by wireless microphones and frequency coordination required to allow multiple microphones to work together at the same site without interference are presented. Finally, it summarizes the regulations in Canada, United States of America, European Union, and Australia for wireless microphones regarding allowable transmitter frequency, power, bandwidth, channel masks, maximum frequency deviation for FM, and spurious emissions.

Wireless microphones are indispensable to modern society. They are used worldwide by professionals and amateurs for such applications as live performances, electronic news gathering, and conference presentations. Due to the radio spectrum crunch, wireless microphones have become a hot topic of discussion and debate among television broadcasters, wireless microphone manufacturers, wireless network operators, and spectrum regulators. In order to keep wireless microphones operating well, it is necessary to give engineers, researchers, and regulators a comprehensive overview of wireless microphones in the aspects of technologies, spectrum requirements, and interference that may be experienced or generated by wireless microphones. This overview is divided into two parts. Part I provides an overview of typical wireless microphone system components, their performance requirements and technical parameters, and the technologies that are used in current wireless microphones. Part II focuses on the spectrum, sources of interference, and regulations surrounding wireless microphones.

Spectrum sensing is an elementary function in cognitive radio designed to monitor the existence of a primary user (PU). To achieve a high rate of detection, most techniques rely on knowledge of prior spectrum patterns, with a trade-off between high computational complexity and long sensing time. On the other hand, blind techniques ignore pattern matching processes to reduce processing time, but their accuracy degrades greatly at low signal-to-noise ratios. To achieve both a high rate of detection and short sensing time, we propose fast spectrum sensing with coordinate system (FSC) - a novel technique that decomposes a spectrum with high complexity into a new coordinate system of salient features and that uses these features in its PU detection process. Not only is the space of a buffer that is used to store information about a PU reduced, but also the sensing process is fast. The performance of FSC is evaluated according to its accuracy and sensing time against six other well-known conventional techniques through a wireless microphone signal based on the IEEE 802.22 standard. FSC gives the best performance overall.

The overarching goal of this project is to develop engineering requirements for future information resources in the ICU. We describe a method and related instrumentation for collecting data about information requirements from a clinical team in situ. Video records are collected using three consumer-grade video cameras, augmented by a wireless microphone system. We combine the video/audio recordings into a single, synchronized record. A transcript is prepared, and then coded by a professional medical records coder, using the SNOMED CT coding scheme. The resulting record represents a coded stream of verbal utterances that is sufficiently precise to support systems engineering analysis.

This paper presents a method for improving Angle-of-Arrival estimation accuracy in multipath environments using a sliding antenna array consisting of only two antennas. The proposed method is then experimentally validated by means of a dual channel Software-Defined Radio receiver and a wireless microphone.

Combining a remote wireless microphone (RM) with hearing aids often improves speech intelligibility for hearing-impaired listeners. However, the head-related acoustic cues that contribute to sound externalization are bypassed when a direct signal link is used. The result is an apparent sound source located within the head of the listener rather than at the talker position. This paper describes using a structural binaural model combined with simulated room reverberation to produce an externalized RM sound source. The binaural model simulates sound propagation around the head, the reflections generated by the pinna, the directional filtering provided by the pinna protrusion from the head, and the ear-canal resonance. The simulated room reverberation uses a two-dimensional virtual image model. The signal-processing blocks used in the binaural model are presented along with examples of the entire system response.

A key technical challenge in TV white spaces is the efficient spectrum usage without interfering with primary users. This paper considers available spectrum discovery scheme using in-band sensing signal to support super Wi-Fi services effectively. The proposed scheme in this paper adopts non-contiguous orthogonal frequency-division multiplexing (NC-OFDM) to utilize the fragmented channel in TV white space due to microphones while this channel cannot be used in IEEE 802.11af. The proposed solution is a novel available spectrum discovery scheme by exploiting the advantages of a sensing signaling. The proposed method achieves considerable improvement in throughput and delay time. The proposed method can use more subcarriers for transmission by applying NC-OFDM in contrast with the conventional IEEE 802.11af standard. Moreover, the increased number of wireless microphones (WMs) hardly affects the throughput of the proposed method because our proposal only excludes some subcarriers used by WMs. Additionally, the proposed method can cut discovery time down to under 10 ms because it can find available channels in real time by exchanging sensing signal without interference to the WM.

プロオーディオの世界で使用されているワイヤレスマイクの歴史は50年以上にも前にさかのぼる。 今やワイヤレスマイクなくして映画作品、 テレビ番組制作はできないと言っても過言ではない。 また、 ライブ・コンサートをはじめとする各種イベント、 選挙の街頭演説、カラオケ、 教育機関の授業と枚挙にいとまないほどに、 ワイヤレスマイクは我々日常生活に深く入り込んでいる。 また、 携帯電話などの電波需要の増加に対応するため、 2011年6月1日の電波法改正で、 A型ワイヤレスマイクで使用していた周波数帯域が携帯電話専用となった。 そして、 2019年4月1日以降は、 全てのA型ワイヤレスマイクが一切使用できなくなる。 これに伴い、 全国の放送局はじめ劇場・ホール等で使用されている約27,000本(2014年2月推定)にも上るワイヤレスマイクは全てこれに則した製品への更新が発生する。 本学、 情報表現学科においても、 「映像ドラマ演習」 はじめ、 様々な演習系授業で、 映像実践においてのワイヤレスマイクの使用は避けられない。 しかし、 使用する場所によって電波がデッドポイント入り受信感度が極端に落ち、 音声がミュートすることがある。 この問題の解消を含め、 筆者の実践を踏まえ、 考察する。, In the professional audio world, wireless microphones have a history of over 50 years. Today, wireless microphones are irreplaceable in the field of film and television production. Even more, wireless microphones are deeply entrenched in our daily lives where they are used in various situations and events such as concerts, presentations, karaoke and classrooms. In order to cope with the rising demand of radio waves resulting from the increase in mobile phones, the frequency band previously used by Type-A wireless microphones was designated for exclusive use with mobile phones in June 1, 2011 in Japan. With this change, all Type-A wireless microphones will become unusable starting April 1, 2019, and as result, approximately 23,000 wireless microphones used in broadcasting stations, theaters, concert halls and other venues around Japan will need to be replaced accordingly. In the Department of Information Expression, wireless microphones are essential as they are used in various exercises and settings in our classrooms. Depending on where a microphone is used, the radio wave may hit a dead point and the reception level of the microphone may drop significantly. In extreme cases, the volume may become completely mute. The following examines solutions for this problem and describes actual measures that were taken.

In this study, the authors consider the problem of signal detection in cognitive radio under the cases of known and unknown noise levels. The analysis is focused on maximum eigenvalue-based method which requires the knowledge of noise variance and the ratio of maximum eigenvalue to the trace of covariance matrix. The analytical formulas are derived for threshold calculation and probability of detection for the considered methods. Performance of the considered methods is measured through simulation analysis. The performance comparison between the considered methods and other existing methods is provided. Simulations based on independent and identically distributed signals and wireless microphone signals are presented to verify the various sensing methods.

In cognitive radio networks, an attacker transmits signals mimicking the characteristics of primary signals, in order to prevent secondary users from transmitting. Such an attack is called primary user emulation (PUE) attack. TV towers and wireless microphones are two main types of primary users in white space. Existing work on PUE attack detection only focused on the first category. For the latter category, primary users are mobile and their transmission power is low. These properties introduce great challenges on PUE detection and existing methods are not applicable. In this paper, we propose a novel method to detect the emulation attack of wireless microphones. We exploit the relationship between RF signals and acoustic information to verify the existence of wireless microphones. The effectiveness of our approach is validated through real-world implementation. Extensive experiments show that our method achieves both false positive rate and false negative rate lower than 0.1 even in a noisy environment.

Noise reduction is an important problem in hands-free speech capture. This paper discusses the use of a remote wireless microphone to aid the reduction of practical non-stationary noise signals. Such a microphone can be placed close to a noise source to provide a reliable estimate of the undesired signal, which can be used by a noise reduction algorithm. The signal from the remote microphone, however, needs to be wirelessly transmitted to the primary device where the noise reduction is performed. As the remote and primary microphones are located on separate devices, their signals need not be synchronously sampled. Moreover, wireless transmission of data is power-intensive, requiring techniques to reduce the transmission bit-rate. This paper discusses transmission of the time domain signal and frequency domain parameters, analyzes their respective trade-offs between performance and robustness to synchronization errors, and presents methods to reduce the transmission bit-rate.

Recent hearing aid systems (HASs) can connect to a wireless microphone worn by the talker of interest. This feature gives the HASs access to a noise-free version of the target signal. In this paper, we address the problem of estimating the target sound direction of arrival (DoA) for a binaural HAS given access to the noise-free content of the target signal. To estimate the DoA, we present a maximum-likelihood framework which takes the shadowing effect of the user's head on the received signals into account by modeling the relative transfer functions (RTFs) between the HAS's microphones. We propose three different RTF models which have different degrees of accuracy and individualization. Furthermore, we show that the proposed DoA estimators can be formulated in terms of inverse discrete Fourier transforms to evaluate the likelihood function computationally efficiently. We extensively assess the performance of the proposed DoA estimators for various DoAs, signal to noise ratios, and in different noisy and reverberant situations. The results show that the proposed estimators improve the performance markedly over other recently proposed “informed” DoA estimator.

Signal detection is a fundamental problem in Cognitive radio. In this paper a new statistical test is proposed based on random data matrix (RDM) for detecting the signals in noise, as opposed to the eigenvalue based tests. Among the many spectrum sensing methods, the RDM method detects the primary users without any prior information. The performance of the test is compared with energy detection (ED), covariance absolute value (CAV) and eigenvalue based algorithms through simulation analysis. This sensing algorithm can be used for very low SNR signal detection without requiring the knowledge of signal, channel and noise. Simulations are based on wireless microphone and identically and independently distributed (iid) signals.

Communication is an essential element in auditorium, gathering, meeting, conference, social gathering etc. A person ought to speak with audience keeping in mind the end goal to create adaptable environment and the communication. Hence the goal of this paper is to present a system to establish effective communication between listener and the speaker. During the utilization of Radio Frequency microphones for communication, an intermediation is required to pass the microphone from one individual to another. This is dull in the event of different inquiries arises at a same time. To stay away from this issue we present the idea where an android cellular phone goes about as receiver. Wi-Fi MIC application needs to be installed in the android phone The application installed on client android phone will act as wireless microphone, thus reducing external hardware devices. The speaker delivering lecture will be server to which all the clients' android phone will be connected through Wi-Fi. This server handles all the clients, giving priority to clients when multiple queries are being raised. Therefore live audio streaming will be carried out in this system.

The effective autonomous cooperation between unmanned aerial vehicles (UAVs) and unmanned ground vehicles (UGVs) has led to an increase in research and development for improving human robot interaction, due to their high potential for achieving successful missions in challenging environments. In this paper, a voice command system is developed to control a UAV and a vision-based approach system controls the UGV in a lead-follower formation to enhance human-robot teams for military defence and rescue missions in urban areas. The proposed voice control framework is executed using open source interface, Robot Operating System (ROS), for aerial vehicle communication supported by spoken dialogues using a wireless microphone, while the UGV detects and tracks a target image captured by the on board low-cost camera from the height of the UAV, using the proposed vision-based approach. The lead-follower formation is based on communication and visibility between the UAV and UGV in a given environment. In addition, a speech recognition module translates voice input to text and the published text is then mapped to the control input of the UAV in order to execute corresponding voice commands. The relative distance and position between the UAV and UGV is estimated from the image received via laptop. UGVs and UAVs have sophisticated design features such as network device communication, navigation, vision sensors and obstacle detection sensors. The UAV is a quad-copter that leads the UGV to accessible areas, while the UGV plays the role of the follower with wheel activators to track and locate the target image on the UAV, given a defined geographic path. The joint behavioural control system of the proposed voice command and vision-based approach is verified by experimental setups that show the UAV has a high voice recognition response and the UGV autonomously tracks the moving UAV in real time operation. As a consequence of the experimental tests, it is expected that the voice control system and vision-based approach developed play an important role in future combat fields.

The revolution in Information Technology(IT) and Artificial Intelligent has provided innovations in diverse kinds of home automation where appliances can be controlled effortlessly and seamlessly .In this paper a review of several home automation technologies was presented and the advantages voice based technology was considered trending among other technologies in terms of flexibility and cost and comfort. In this paper, the conceptual prototype design model and the circuit diagram for controlling multiple home appliances at the same time were presented. The model uses Arduino Uno microcontroller system to control multiple home electrical or electronic appliances with maximum of 5 volts. The design transmits voice command through user wireless microphone (attached to the user cloth) to the intelligence micro controller via graphical user interphase (GUI). The control software developed for the prototype model was implemented using C# programming language, Microsoft Visual Studio.Net and Microsoft Speech recognition system.

In this paper, we propose the technical considerations for the rules on television white space(TVWS), that becomes a center of public notice, in terms of incumbent services protection, technical requirements of television band devices (TVBDs) and administration of TV band databases. The basic idea behind the proposed method is based on the perfect protection of incumbent services such as digital television and wireless microphone. It has been found that FCC`s approach has the advantage that a list of available channels and transmission power are provided for TVBDs faster than Ofcom`s approach that requires DB to perform the algorithm for determining frequencies and transmission power for TVBDs everytime DB is asked from TVBDs.

This article describes an experimental testbed that combines wireless microphone sensors with a web-based terrestrial digital video broadcasting geolocation database and a program making and special events spectrumbooking platform. Software defined radio and Internet technologies are the enabling tools in use. The key sensing techniques are identified and implemented. Test trials in a real scenario have shown that the platform was able to receive information from a DVB-T geolocation database and a PMSE spectrum-booking platform, updating the list of vacant channels with blind sensing techniques. The proposed method has shown capabilities to protect primary users of interferences from secondary users of the spectrum.

Summary 1. Using arrays of microphones, biologists can monitor the position of free-living animals based on the sounds they produce. Microphone array technology exploits differences in sound arrival times at each microphone to calculate an animal’s position. This technology provides new opportunities for studying animal ecology and behaviour and has many advantages over tracking technologies that require capturing animals and fitting them with external devices, or technologies that focus on one individual in isolation of the activities of nearby animals. 2. The efficacy of microphone arrays for triangulating the position of wild animals has been established through previous studies. Yet widespread use of microphone array technology has been limited by many factors: arrays are expensive, custom manufactured, and cumbersome. Consequently, microphone arrays are used infrequently, in spite of their transformative potential for studying animal ecology and behaviour. 3. We conducted a field test of a new wireless microphone array system that has multiple advantages over previous systems: it is relatively inexpensive, commercially available, includes an integrated global positioning system (GPS) for time-synchronizing microphones, and it is small enough to fit in a backpack. We set up an array of four stereo recorders (each with a pair of stereo microphones) at 12 sites and tested the system’s accuracy for estimating the location of loudspeakers broadcasting 25 types of bird, mammal and frog sounds. 4. We found that this system produced accurate location estimates based on multi-channel recordings of many types of acoustic signals. The average location accuracy was 1·87 ± 0·13 m, on par with cable-based microphone array systems. Location accuracy was significantly higher when the recorders were closer together and when sounds were broadcast inside the area bounded by the microphones. Accuracy tended to be higher in field vs. forest habitats. 5. We discuss how this system may be used to enhance studies of animal ecology and behaviour across a wide range of contexts. As with previous arrays, this system will allow researchers to monitor animals that produce distinctive acoustic signals. In contrast to previous microphone arrays, this system is affordable, portable and commercially available. Consequently, this system stands to dramatically enhance research on wild, free-living animals.