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Microphone

About: Microphone is a research topic. Over the lifetime, 39999 publications have been published within this topic receiving 337352 citations. The topic is also known as: mic & mike.


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Patent
17 Apr 2008
TL;DR: In this paper, the location of the microphones roughly coincides with the position of ears on a human body, which creates a mobile robot that more effectively simulates the tele-presence of an operator of the system.
Abstract: A remote controlled robot system that includes a robot and a remote control station. The robot includes a binaural microphone system that is coupled to a speaker system of the remote control station. The binaural microphone system may include a pair of microphones located at opposite sides of a robot head. the location of the microphones roughly coincides with the location of ears on a human body. Such microphone location creates a mobile robot that more effectively simulates the tele-presence of an operator of the system. The robot may include two different microphone systems and the ability to switch between systems. For example, the robot may also include a zoom camera system and a directional microphone. The directional microphone may be utilized to capture sound from a direction that corresponds to an object zoomed upon by the camera system.

577 citations

Journal ArticleDOI
TL;DR: Alternative spatial sampling schemes for the positioning of microphones on a sphere are presented, and the errors introduced by finite number of microphones, spatial aliasing, inaccuracies in microphone positioning, and measurement noise are investigated both theoretically and by using simulations.
Abstract: Spherical microphone arrays have been recently studied for sound-field recordings, beamforming, and sound-field analysis which use spherical harmonics in the design. Although the microphone arrays and the associated algorithms were presented, no comprehensive theoretical analysis of performance was provided. This work presents a spherical-harmonics-based design and analysis framework for spherical microphone arrays. In particular, alternative spatial sampling schemes for the positioning of microphones on a sphere are presented, and the errors introduced by finite number of microphones, spatial aliasing, inaccuracies in microphone positioning, and measurement noise are investigated both theoretically and by using simulations. The analysis framework can also provide a useful guide for the design and analysis of more general spherical microphone arrays which do not use spherical harmonics explicitly.

522 citations

Proceedings ArticleDOI
Chunyi Peng1, Guobin Shen1, Yongguang Zhang1, Yanlin Li1, Kun Tan1 
06 Nov 2007
TL;DR: The design, implementation, and evaluation of BeepBeep is presented, a high-accuracy acoustic-based ranging system that operates in a spontaneous, ad-hoc, and device-to-device context without leveraging any pre-planned infrastructure.
Abstract: We present the design, implementation, and evaluation of BeepBeep, a high-accuracy acoustic-based ranging system. It operates in a spontaneous, ad-hoc, and device-to-device context without leveraging any pre-planned infrastructure. It is a pure software-based solution and uses only the most basic set of commodity hardware -- a speaker, a microphone, and some form of device-to-device communication -- so that it is readily applicable to many low-cost sensor platforms and to most commercial-off-the-shelf mobile devices like cell phones and PDAs. It achieves high accuracy through a combination of three techniques: two-way sensing, self-recording, and sample counting. The basic idea is the following. To estimate the range between two devices, each will emit a specially-designed sound signal ("Beep") and collect a simultaneous recording from its microphone. Each recording should contain two such beeps, one from its own speaker and the other from its peer. By counting the number of samples between these two beeps and exchanging the time duration information with its peer, each device can derive the two-way time of flight of the beeps at the granularity of sound sampling rate. This technique cleverly avoids many sources of inaccuracy found in other typical time-of-arrival schemes, such as clock synchronization, non-real-time handling, software delays, etc. Our experiments on two common cell phone models have shown that we can achieve around one or two centimeters accuracy within a range of more than ten meters, despite a series of technical challenges in implementing the idea.

519 citations

01 Jan 1994
TL;DR: An extensive set of head-related transfer function (HRTF 1) measurements of a KEMAR dummy head microphone have recently been completed, which consist of the left and right ear impulse responses from a Realistic Optimus Pro 7 loudspeaker mounted 1.4 meters from the K EMAR.
Abstract: An extensive set of head-related transfer function (HRTF 1) measurements of a KEMAR dummy head microphone has recently been completed. The measurements consist of the left and right ear impulse responses from a Realistic Optimus Pro 7 loudspeaker mounted 1.4 meters from the KEMAR. Maximum length (ML) pseudo-random binary sequences were used to obtain the impulse responses at a sampling rate of 44.1 kHz. In total, 710 diierent positions were sampled at elevations from-40 degrees to +90 degrees. Also measured were the impulse response of the speaker in free eld and several headphones placed on the KEMAR. This data is being made available to the research community on the Internet via anonymous FTP and the World Wide Web. 1 Measurement technique Measurements were made using a Macintosh Quadra computer equipped with an Audiomedia II DSP card, which has 16-bit stereo A/D and D/A converters that operate at a 44.1 kHz sampling rate. One of the audio output channels was sent to an ampliier which drove a Realistic Optimus Pro 7 loudspeaker. This is a small two way loudspeaker with a 4 inch woofer and 1 inch tweeter. The KEMAR, Knowles Electronics model DB-4004, was equipped with model DB-061 left pinna, model DB-065 (large red) right pinna, Etymotic ER-11 microphones, and Etymotic ER-11 preampliiers. The outputs of the microphone preampliiers were connected to the stereo inputs of the Audiomedia card. From the standpoint of the Audiomedia card, a signal sent to the audio outputs results in a corresponding signal appearing at the audio inputs. Measuring the impulse response of this system yields the impulse response of the combined system consisting of the Audiomedia D/A and A/D converters and anti-alias lters, the ampliier, the speaker, the room in which the measurements are made, and most importantly, the response of the KEMAR with its associated microphones and preamps. We can avoid interference due to room reeections by ensuring that any reeections occur well after the head response time, which is several milliseconds. We can compensate for a non-uniform speaker response by measuring the speaker response separately and creating an inverse lter. The inverse lter, when applied to an HRTF measurement, equalizes the speaker response to be at. 1 In this document, we use the acronym HRTF to refer to head related impulse responses. The impulse response and transfer function are related in the obvious way by the Fourier transform.

505 citations

Patent
24 Jul 2008
TL;DR: In this article, a speech recognition system consisting of a light element, a power control switch, a controller, a microphone, a speech recognizer coupled to the microphone for recognizing speech input signals and transmitting recognition results to the controller, and a speech synthesizer coupled with the controller for generating synthesized speech was presented.
Abstract: The present invention includes a speech recognition system comprising a light element, a power control switch, the power control switch varying the power delivered to the light element, a controller, a microphone, a speech recognizer coupled to the microphone for recognizing speech input signals and transmitting recognition results to the controller, and a speech synthesizer coupled to the controller for generating synthesized speech, wherein the controller varies the power to the light element in accordance with the recognition results received from the speech recognizer. Embodiments of the invention may alternatively include a low power wake up circuit. In another embodiment, the present invention is a method of controlling a device by voice commands.

495 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
2023493
2022985
2021670
20201,638
20191,955
20182,056