Prof. Hanna Bogucka, head of the Department of Wireless Communications at the Poznań University of Technology, discusses unnecessary inhibitions, the usefulness of microphones, and the links between people and technology.
This paper presents and compares microphone calibration methods for the simultaneous calibration of small electret microphones in a wave guide. The microphones are simultaneously calibrated to a reference microphone both in amplitude and phase. The calibration procedure is formulated on the basis of the damped plane wave propagation equation, from which the acoustics field along the wave guide is predicted, using several reference measurements. Different calibration models are presented and the methods were found to be sensitive to the formulation, as well as to the number of free parameters used during the reconstruction of the wave-field. The wave guide model based on five free parameters was found to be the preferred method for this type of calibration procedure.
Whenever the recording engineer uses stereo microphone techniques, he/she has to consider a recording angle resulting from the positioning of microphones relative to sound sources, besides other acoustic factors. The recording angle, the width of a captured acoustic scene and the properties of a particular microphone technique are closely related. We propose a decision supporting method, based on the mapping of the actual position of a sound source to its position in the reproduced acoustic scene. This research resulted in a set of localisation curves characterising four most popular stereo microphone techniques. The curves were obtained by two methods: calculation, based on appropriate engineering formulae, and experiment consisting in the recording of sources and estimation of the perceived position in listening tests. The analysis of curves brings several conclusions important in the recording practice.
The development of digital microphones and loudspeakers adds new and interesting possibilities of their applications in different fields, extended from industrial, medical to consumer audio markets. One of the rapidly growing field of applications is mobile multimedia, such as mobile phones, digital cameras, laptop and desktop PCs, etc. The advances have also been made in digital audio, particularly in direct digital transduction, so it is now possible to create the all-digital audio recording and reproduction chains potentially having several advantages over existing analog systems.
The objective of the study is to assess the hearing performance of cochlear implant users in three device microphone configurations: omni-directional, directional, and beamformer (BEAMformer two-adaptive noise reduction system), in localization and speech perception tasks in dynamically changing listening environments. Seven cochlear implant users aided with Cochlear CM-24 devices with Freedom speech processor participated in the study. For the localization test in quiet and in background noise, subjects demonstrated significant differences between different microphone settings. Confusion matrices showed that in about 70% cases cochlear implant subjects correctly localized sounds within a horizontal angle of 30-40◦ (±1◦ loudspeaker apart from signal source). However localization in noise was less accurate as shown by a large number of considerable errors in localization in the confusion matrices. Average results indicated no significant difference between three microphone configurations. For speech presented from the front 3 dB SNR improvements in speech intelligibility in three subjects can be observed for beamforming system compared to directional and omni-directional microphone settings. The benefits of using different microphone settings in cochlear implant devices in dynamically changing listening conditions depend on the particular sound environment
Passive noise reduction means are commonly used to reduce noise in the industry but, unfortunately, their effectiveness is poor in the low frequency range. By applying active structural acoustic control to the enclosure walls significant improvement of the insulating properties in this frequency range can be achieved. In this paper a model of double panel structure with ASAC is presented. The structure consists of two aluminium plates separated by an air gap. Two inertial magnetoelectric actuators and two piezoceramic MFC sensors were used for controlling the structure. A multichannel FxLMS algorithm with virtual error microphone technique is used as a control algorithm. The signal of a virtual error microphone is extrapolated basing on signals from MFC sensors. Performance of this actively controlled structure for tonal signals at selected frequencies is presented in the article. During the study, a double panel structure was mounted on one wall of sound insulating enclosure located in an acoustic chamber. During the measurements local and global reduction of noise test signal was investigated.
The noise of motor vehicles is one of the most important problems as regards to pollution on main roads. However, this unpleasant characteristic could be used to determine vehicle speed by external observers. Building on this idea, the present study investigates the capabilities of a microphone array system to identify the position and velocity of a vehicle travelling on a previously established route. Such linear microphone array has been formed by a reduced number of microphones working at medium frequencies as compared to industrial microphone arrays built for location purposes, and operates with a processing algorithm that ultimately identifies the noise source location and reduces the error in velocity estimation
The aim of this paper is to describe the process of choosing the best surround microphone technique for recording of choir with an instrumental ensemble. First, examples of multichannel microphone techniques including those used in the recording are described. Then, the assumptions and details of music recording in Radio Gdansk Studio are provided as well as the process of mixing of the multichannel recording. The extensive subjective tests were performed employing a group of sound engineers and students in order to find the most preferable recording techniques. Because the final recording is based on the mix of "direct/ambient" and "direct-sound all-around" approaches, a subjective quality evaluation was conducted and on this basis the best rated multichannel techniques were chosen. The results show that listeners might consider different factors when choosing the best rated multichannel techniques in separate tasks, as different systems were chosen in the two tests.
This paper gives a detailed electroacoustic study of a new generation of monolithic CMOS micromachined electrodynamic microphone, made with standard CMOS technology. The monolithic integration of the mechanical sensor with the electronics using a standard CMOS process is respected in the design, which presents the advantage of being inexpensive while having satisfactory performance. The MEMS microphone structure consists mainly of two planar inductors which occupy separate regions on substrate. One inductor is fixed; the other can exercise out-off plane movement. Firstly, we detail the process flow, which is used to fabricate our monolithic microphone. Subsequently, using the analogy between the three different physical domains, a detailed electro-mechanical-acoustic analogical analysis has been performed in order to model both frequency response and sensitivity of the microphone. Finally, we show that the theoretical microphone sensitivity is maximal for a constant vertical position of the diaphragm relative to the substrate, which means the distance between the outer and the inner inductor. The pressure sensitivity, which is found to be of the order of a few tens of μV/Pa, is flat within a bandwidth from 50 Hz to 5 kHz.