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Logarithmic Image Sensor for Wide Dynamic Range Stereo Vision System
Logarithmic Image Sensor For Wide Dynamic Range Stereo Vision System Christian Bouvier, Yang Ni New Imaging Technologies SA, 1 Impasse de la Noisette, BP 426, 91370 Verrieres le Buisson France Tel: +33 (0)1 64 47 88 58 [email protected], [email protected] NEG PIXbias COLbias Abstract—Stereo vision is the most universal way to get 3D information passively. The fast progress of digital image G1 processing hardware makes this computation approach realizable Vsync G2 Vck now. It is well known that stereo vision matches 2 or more Control Amp Pixel Array - Scan Offset images of a scene, taken by image sensors from different points RST - 1280x720 of view. Any information loss, even partially in those images, will V Video Out1 drastically reduce the precision and reliability of this approach. Exposure Out2 In this paper, we introduce a stereo vision system designed for RD1 depth sensing that relies on logarithmic image sensor technology. RD2 FPN Compensation The hardware is based on dual logarithmic sensors controlled Hsync and synchronized at pixel level. This dual sensor module provides Hck H - Scan high quality contrast indexed images of a scene. This contrast indexed sensing capability can be conserved over more than 140dB without any explicit sensor control and without delay. Fig. 1. Sensor general structure. It can accommodate not only highly non-uniform illumination, specular reflections but also fast temporal illumination changes. Index Terms—HDR, WDR, CMOS sensor, Stereo Imaging. the details are lost and consequently depth extraction becomes impossible. The sensor reactivity will also be important to prevent saturation in changing environments. -
AEA-KU5A-Owners-Manual-Rev1.Pdf
AEA KU5A OWNER’S MANUAL ACTIVE CARDIOID END-ADDRESS RIBBON MIC WELCOME We congratulate you on your purchase of the AEA KU5A ribbon microphone and welcome you to the AEA family. Never before has a ribbon mic delivered such directionality and uncompromised tonality in a package so fit for both studio and live settings. The KU5A is ideal for studio and live applications where both superior rejection and classic ribbon tonality are paramount. Whether recording a vocal, guitar, or even a snare drum, the KU5A delivers brilliant, focused sound where other conventional ribbons can’t. From close range the KU5A retains the hefty low end and pronounced midrange one expects of AEA ribbons and with manageable proximity effect. Vocalists can sing directly into the grille of the KU5A because its interior components are the most protected of any in the AEA lineup. The rugged KU5A is equipped with active electronics, so it’s suited or any preamp in the studio and on the road. Your KU5A microphone is 100% handcrafted in Pasadena, CA. AEA is an independent, family owned company with a small crew of skilled technicians – many of whom are themselves, musicians. We manufacture all our ribbon microphones and preamps with locally sourced parts. We hope your microphone will capture many magical performances that touch the heart. This manual will help ensure that you get the best sound and longevity from your new microphone. Please become part of the AEA community by sharing your experiences via e-mail, phone or social media. Wes Dooley President of AEA 2 CONTENTS 2 WELCOME 4 INTRODUCTION 4 SUPPORT 5 GENERAL GUIDELINES 7 APPLICATION ADVICE 10 SPECIFICATIONS 3 INTRODUCTION The KU5A is an end-address, phantom-powered, supercardioid ribbon microphone. -
Receiver Dynamic Range: Part 2
The Communications Edge™ Tech-note Author: Robert E. Watson Receiver Dynamic Range: Part 2 Part 1 of this article reviews receiver mea- the receiver can process acceptably. In sim- NF is the receiver noise figure in dB surements which, taken as a group, describe plest terms, it is the difference in dB This dynamic range definition has the receiver dynamic range. Part 2 introduces between the inband 1-dB compression point advantage of being relatively easy to measure comprehensive measurements that attempt and the minimum-receivable signal level. without ambiguity but, unfortunately, it to characterize a receiver’s dynamic range as The compression point is obvious enough; assumes that the receiver has only a single a single number. however, the minimum-receivable signal signal at its input and that the signal is must be identified. desired. For deep-space receivers, this may be COMPREHENSIVE MEASURE- a reasonable assumption, but the terrestrial MENTS There are a number of candidates for mini- mum-receivable signal level, including: sphere is not usually so benign. For specifi- The following receiver measurements and “minimum-discernable signal” (MDS), tan- cation of general-purpose receivers, some specifications attempt to define overall gential sensitivity, 10-dB SNR, and receiver interfering signals must be assumed, and this receiver dynamic range as a single number noise floor. Both MDS and tangential sensi- is what the other definitions of receiver which can be used both to predict overall tivity are based on subjective judgments of dynamic range do. receiver performance and as a figure of merit signal strength, which differ significantly to compare competing receivers. -
For the Falcon™ Range of Microphone Products (Ba5105)
Technical Documentation Microphone Handbook For the Falcon™ Range of Microphone Products Brüel&Kjær B K WORLD HEADQUARTERS: DK-2850 Nærum • Denmark • Telephone: +4542800500 •Telex: 37316 bruka dk • Fax: +4542801405 • e-mail: [email protected] • Internet: http://www.bk.dk BA 5105 –12 Microphone Handbook Revision February 1995 Brüel & Kjær Falcon™ Range of Microphone Products BA 5105 –12 Microphone Handbook Trademarks Microsoft is a registered trademark and Windows is a trademark of Microsoft Cor- poration. Copyright © 1994, 1995, Brüel & Kjær A/S All rights reserved. No part of this publication may be reproduced or distributed in any form or by any means without prior consent in writing from Brüel & Kjær A/S, Nærum, Denmark. 0 − 2 Falcon™ Range of Microphone Products Brüel & Kjær Microphone Handbook Contents 1. Introduction....................................................................................................................... 1 – 1 1.1 About the Microphone Handbook............................................................................... 1 – 2 1.2 About the Falcon™ Range of Microphone Products.................................................. 1 – 2 1.3 The Microphones ......................................................................................................... 1 – 2 1.4 The Preamplifiers........................................................................................................ 1 – 8 1 2. Prepolarized Free-field /2" Microphone Type 4188....................... 2 – 1 2.1 Introduction ................................................................................................................ -
Signal-To-Noise Ratio and Dynamic Range Definitions
Signal-to-noise ratio and dynamic range definitions The Signal-to-Noise Ratio (SNR) and Dynamic Range (DR) are two common parameters used to specify the electrical performance of a spectrometer. This technical note will describe how they are defined and how to measure and calculate them. Figure 1: Definitions of SNR and SR. The signal out of the spectrometer is a digital signal between 0 and 2N-1, where N is the number of bits in the Analogue-to-Digital (A/D) converter on the electronics. Typical numbers for N range from 10 to 16 leading to maximum signal level between 1,023 and 65,535 counts. The Noise is the stochastic variation of the signal around a mean value. In Figure 1 we have shown a spectrum with a single peak in wavelength and time. As indicated on the figure the peak signal level will fluctuate a small amount around the mean value due to the noise of the electronics. Noise is measured by the Root-Mean-Squared (RMS) value of the fluctuations over time. The SNR is defined as the average over time of the peak signal divided by the RMS noise of the peak signal over the same time. In order to get an accurate result for the SNR it is generally required to measure over 25 -50 time samples of the spectrum. It is very important that your input to the spectrometer is constant during SNR measurements. Otherwise, you will be measuring other things like drift of you lamp power or time dependent signal levels from your sample. -
High Dynamic Range Ultrasound Imaging
The final publication is available at http://link.springer.com/article/10.1007/s11548-018-1729-3 Int J CARS manuscript No. (will be inserted by the editor) High Dynamic Range Ultrasound Imaging Alperen Degirmenci · Douglas P. Perrin · Robert D. Howe Received: 26 January 2018 Abstract Purpose High dynamic range (HDR) imaging is a popular computational photography technique that has found its way into every modern smartphone and camera. In HDR imaging, images acquired at different exposures are combined to increase the luminance range of the final image, thereby extending the limited dynamic range of the camera. Ultrasound imaging suffers from limited dynamic range as well; at higher power levels, the hyperechogenic tissue is overexposed, whereas at lower power levels, hypoechogenic tissue details are not visible. In this work, we apply HDR techniques to ultrasound imaging, where we combine ultrasound images acquired at different power levels to improve the level of detail visible in the final image. Methods Ultrasound images of ex vivo and in vivo tissue are acquired at different acoustic power levels and then combined to generate HDR ultrasound (HDR-US) images. The performance of five tone mapping operators is quantitatively evaluated using a similarity metric to determine the most suitable mapping for HDR-US imaging. Results The ex vivo and in vivo results demonstrated that HDR-US imaging enables visualizing both hyper- and hypoechogenic tissue at once in a single image. The Durand tone mapping operator preserved the most amount of detail across the dynamic range. Conclusions Our results strongly suggest that HDR-US imaging can improve the utility of ultrasound in image-based diagnosis and procedure guidance. -
Bachelor Thesis
BACHELOR THESIS Relationship Between Distance and Microphone Directivity in a Speaker Booth Björn Samuelsson 2015 Bachelor of Arts Audio Engineering Luleå University of Technology Department of Arts, Communication and Education Relationship between distance and microphone directivity in a speaker booth Abstract A study was made to investigate trained listeners preferences regarding a microphone's directivity and distance from a speaker when listening to a voice recorded in a speaker booth. Two directivities (omnidirectional and cardioid) and three distances (5, 25 and 52 cm) were tested on 19 subjects in a multiple comparison listening test regarding their preference and five other attributes taken from earlier studies. The result showed that an omnidirectional microphone placed close to a source might be preferable over a cardioid at the same distance or a cardioid placed a bit away from the source. The omnidirectional microphone however tends to perform more even at different distances compared to a cardioid, which makes it easier for a engineer to find a good microphone position for it. Table of Content Acknowledgements..........................................................................................................3 1. Introduction.....................................................................................................................4 1.1. Aim............................................................................................................................4 1.2. Research question....................................................................................................4 -
Preview Only
AES-6id-2006 (r2011) AES information document for digital audio - Personal computer audio quality measurements Published by Audio Engineering Society, Inc. Copyright © 2006 by the Audio Engineering Society Preview only Abstract This document focuses on the measurement of audio quality specifications in a PC environment. Each specification listed has a definition and an example measurement technique. Also included is a detailed description of example test setups to measure each specification. An AES standard implies a consensus of those directly and materially affected by its scope and provisions and is intended as a guide to aid the manufacturer, the consumer, and the general public. An AES information document is a form of standard containing a summary of scientific and technical information; originated by a technically competent writing group; important to the preparation and justification of an AES standard or to the understanding and application of such information to a specific technical subject. An AES information document implies the same consensus as an AES standard. However, dissenting comments may be published with the document. The existence of an AES standard or AES information document does not in any respect preclude anyone, whether or not he or she has approved the document, from manufacturing, marketing, purchasing, or using products, processes, or procedures not conforming to the standard. Attention is drawn to the possibility that some of the elements of this AES standard or information document may be the subject of patent rights. AES shall not be held responsible for identifying any or all such patents. This document is subject to periodicwww.aes.org/standards review and users are cautioned to obtain the latest edition and printing. -
Proximity Effect Detection for Directional Microphones
Audio Engineering Society Convention Paper 8475 Presented at the 131st Convention 2011 October 20–23 New York, USA This Convention paper was selected based on a submitted abstract and 750-word precis that have been peer reviewed by at least two qualified anonymous reviewers. The complete manuscript was not peer reviewed. This convention paper has been reproduced from the author’s advance manuscript without editing, corrections, or consideration by the Review Board. The AES takes no responsibility for the contents. Additional papers may be obtained by sending request and remittance to Audio Engineering Society, 60 East 42nd Street, New York, New York 10165-2520, USA; also see www.aes.org. All rights reserved. Reproduction of this paper, or any portion thereof, is not permitted without direct permission from the Journal of the Audio Engineering Society. Proximity effect detection for directional microphones Alice Clifford1 and Joshua D. Reiss1 1Queen Mary University of London, London, E1 4NS, UK Correspondence should be addressed to Alice Clifford ([email protected]) ABSTRACT The proximity effect in directional microphones is characterised by an undesired boost in low frequency energy as the source to microphone distance decreases. Traditional methods for reducing the proximity effect use a high pass filter to cut low frequencies which alter the tonal characteristics of the sound and are not dependent on the input source. This paper proposes an intelligent approach to detect the proximity effect in a single capsule directional microphone in real time. The low frequency boost is detected by analysing the spectral flux of the signal over a number of bands over time. -
Understanding Microphone Sensitivity by Jerad Lewis
Understanding Microphone Sensitivity By Jerad Lewis Sensitivity, the ratio of the analog output voltage or digital output value to the input pressure, is a key specification of any microphone. Mapping units in the acoustic domain to units in the electrical domain, it determines the magnitude of the microphone output signal given a known input. This article will discuss the distinction in sensitivity specifications between analog and digital microphones, how to choose a microphone with the best sensitivity for the application, and why adding a bit (or more) of digital gain can enhance the microphone signal. Analog vs. Digital Microphone sensitivity is typically measured with a 1 kHz sine wave at a 94 dB sound pressure level (SPL), or 1 pascal (Pa) pressure. The magnitude of the analog or digital output signal from the microphone with that input stimulus is a measure of its sensitivity. This reference point is but one characteristic of the microphone, by no means the whole story of its performance. The sensitivity of an analog microphone is straightforward and easy to understand. Typically specified in logarithmic units of dBV (decibels with respect to 1 V), it tells how many volts the output signal will be for a given SPL. For an analog microphone, sensitivity, in linear units of mV/Pa, can be expressed logarithmically in decibels: Sensitivity = × mV / Pa Sensitivity dBV 20 log10 Output REF where OutputREF is the 1000 mV/Pa (1 V/Pa) reference output ratio. Given this information, with the appropriate preamplifier gain, the microphone signal level can be easily matched to the desired input level of the rest of the circuit or system. -
A History of Audio Effects
applied sciences Review A History of Audio Effects Thomas Wilmering 1,∗ , David Moffat 2 , Alessia Milo 1 and Mark B. Sandler 1 1 Centre for Digital Music, Queen Mary University of London, London E1 4NS, UK; [email protected] (A.M.); [email protected] (M.B.S.) 2 Interdisciplinary Centre for Computer Music Research, University of Plymouth, Plymouth PL4 8AA, UK; [email protected] * Correspondence: [email protected] Received: 16 December 2019; Accepted: 13 January 2020; Published: 22 January 2020 Abstract: Audio effects are an essential tool that the field of music production relies upon. The ability to intentionally manipulate and modify a piece of sound has opened up considerable opportunities for music making. The evolution of technology has often driven new audio tools and effects, from early architectural acoustics through electromechanical and electronic devices to the digitisation of music production studios. Throughout time, music has constantly borrowed ideas and technological advancements from all other fields and contributed back to the innovative technology. This is defined as transsectorial innovation and fundamentally underpins the technological developments of audio effects. The development and evolution of audio effect technology is discussed, highlighting major technical breakthroughs and the impact of available audio effects. Keywords: audio effects; history; transsectorial innovation; technology; audio processing; music production 1. Introduction In this article, we describe the history of audio effects with regards to musical composition (music performance and production). We define audio effects as the controlled transformation of a sound typically based on some control parameters. As such, the term sound transformation can be considered synonymous with audio effect. -
Fast Multiplierless Approximations of the DCT with the Lifting Scheme Jie Liang, Student Member, IEEE, and Trac D
3032 IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 49, NO. 12, DECEMBER 2001 Fast Multiplierless Approximations of the DCT With the Lifting Scheme Jie Liang, Student Member, IEEE, and Trac D. Tran, Member, IEEE Abstract—In this paper, we present the design, implementation, the sparse factorizations of the DCT matrix [12]–[17], and and application of several families of fast multiplierless approx- many of them are recursive [12], [14], [16], [17]. Besides imations of the discrete cosine transform (DCT) with the lifting one-dimensional (1-D) algorithms, two-dimensional (2-D) scheme called the binDCT. These binDCT families are derived from Chen’s and Loeffler’s plane rotation-based factorizations of DCT algorithms have also been investigated extensively [6], the DCT matrix, respectively, and the design approach can also [18]–[21], generally leading to less computational complexity be applied to a DCT of arbitrary size. Two design approaches are than the row-column application of the 1-D methods. However, presented. In the first method, an optimization program is defined, the implementation of the direct 2-D DCT requires much more and the multiplierless transform is obtained by approximating effort than that of the separable 2-D DCT. its solution with dyadic values. In the second method, a general lifting-based scaled DCT structure is obtained, and the analytical The theoretical lower bound on the number of multiplications values of all lifting parameters are derived, enabling dyadic required for the 1-D eight-point DCT has been proven to be 11 approximations with different accuracies. Therefore, the binDCT [22], [23]. In this sense, the method proposed by Loeffler et al.