Microphone

Now a day, peoples have multiple bank accounts so money transactions play a vital role in the nature of trade. Today, ATMs and Credit cards are used for this purpose, the authentication of these transactions are unsecure. To overcome this shortcoming of money transactions, we proposes the idea of using fingerprints of customers as login multiple banking password in place of traditional pin number. Here, if the fingerprint is recognized, then it display the multiple banking screen. Then we can choose the bank which we need for transaction. The remaining feature are same as i.e., a reference fingerprint of the nominee or a close family member of the customer can be used if the customer is not available in case of emergencies. This proposed business model helps the society, mainly the rural people, by enhancing the security using Fingerprint recognition in Digital image processing. As the fingerprint of every person is unique and unchangeable, this biometric feature is used over the others

This paper presents an application-oriented approach to Acoustic Source Localization using a coincident microphone array. Multiple processing blocks are presented to generate a reactive, yet stable Direction of Arrival estimation tuned toward speaker tracking. Building on an energy based scanning method, individual characteristics, such as sound field directivity and static sound source positions are used for adaptive smoothing of the detected angle. The methods and resulting performance gain are discussed for the individual components of the algorithm. Objective performance is evaluated using simulated and recorded data. Audio quality is assessed using listening tests, which show a significant increase in subjective sound quality, noise suppression, and speech intelligibility when combining the tracker with a beamforming algorithm for coincident microphone arrays.

Performance of axial fan is found to reduce drastically when instability is
encountered during its operation. Performance of an axial fan is severely impaired by many factors mostly related to system instabilities due to rotating stall and surge
phenomenon experienced during its operation. The present work involves measuring the time domain sound signal in ducted axial fan under stable region at throttle positions 5 cm from the casing. Objective of the experiment is to measure the time domain sound signal in terms of decibel and comprehend the sound Characteristics in ducted axial fan by using sound spectrum analyser. Different types of time domain sound signals have been measured under stable region at throttle position 5 cm from the casing for different rotor speed and different graphs are plotted for ducted axial fan

This paper analyses the influence of microphone cavity geometry on beamforming measurements with a turbulent boundary layer present on the microphone array. A 16- microphone array was tested in the anechoic open-jet wind tunnel of the Delft University of Technology. The array was placed on a flat plate mounted flush with the exit nozzle of the wind tunnel. Microphones were installed in three different cavity geometries along with a flush mounted microphone array which was used as a baseline for comparison. The geometries include a chamfered-cylindrical hard-plastic cavity, a chamfered-cylindrical cavity made of melamine foam, and a chamfered-cylindrical cavity with star-shaped protrusions, also made of melamine. The recessed cavities were covered with a 0.026 mm gauge steelwire cloth. A speaker emitting white noise outside of the flow was employed as a sound source. Different flow velocities at the same sound power level for the speaker (and hence different signal-to-noise ratios) w...

This study looks at two ways of extracting a glottal waveform from recorded speech. One way is to inverse filter the flow at the mouth. Another is to inverse filter the microphone signal. Theoretically, the microphone signal is considered to be the equivalent of a first order differentiation of the flow signal recorded at the lips. Recording the oral airflow is more complicated than the recording of a microphone signal, as it requires the use of a mask, with constant adjustments during the recording. Recording of the microphone signal is more straightforward for the experimenter and less intrusive for the subject. If the two inverse filtering procedures can be shown to produce similar glottal flow waveforms for both types of recorded speech, this would support the use of only the microphone signal for those types of glottal flow analysis where the DC component of the flow is not essetial, making voice source analysis applicable in less specialised situations. In this study, we used ...

Objective measurements using a sinusoidal sweep show that microphone angle has little effect on the frequency response of a guitar amplifier recording [1]. However, anecdotal evidence suggests that alterations to the microphone angle hold merit when recording ecologically valid sound sources. An ABX listening experiment was conducted with 20 participants to investigate whether microphone angles of 0, 30, and 60 degrees were audibly different to this cohort. Both dynamic and ribbon microphones were used and the loudness normalized guitar recordings were presented in solo and within a music mix. The experimental results suggest that microphone angles did not generate any perceivable changes to this cohort on this program material.

International audienceThis article traces the history of the Locustream Open Microphone Project, a network of open microphones that has streamed live soundscapes from locations around the globe since 2005. The author describes the human, collective , and technological aspects of this project as well as the creative research it has engendered

This article introduces and provides an assessment of a spatial-filtering algorithm based on two closely-spaced (∼1 cm) microphones in a behind-the-ear shell. The evaluated spatial-filtering algorithm used fast (∼10 ms) temporal-spectral analysis to determine the location of incoming sounds and to enhance sounds arriving from straight ahead of the listener. Speech reception thresholds (SRTs) were measured for eight cochlear implant (CI) users using consonant and vowel materials under three processing conditions: An omni-directional response, a dipole-directional response, and the spatial-filtering algorithm. The background noise condition used three simultaneous time-reversed speech signals as interferers located at 90°, 180°, and 270°. Results indicated that the spatial-filtering algorithm can provide speech reception benefits of 5.8 to 10.7 dB SRT compared to an omni-directional response in a reverberant room with multiple noise sources. Given the observed SRT benefits, coupled wi...